Chapter An Introduction to 1 Traditional Telephony …...An Introduction to Traditional Telephony and Cisco Unified Communications THE FOLLOWING CVOICE EXAM OBJECTIVES ARE COVERED

An Introduction to Traditional Telephony and Cisco Unified Communications

THE FOLLOWING CVOICE EXAM OBJECTIVES ARE COVERED IN THIS CHAPTER:

Describe the components of a gateway.

■ Describe the function of gateways.

Describe a dial plan.

■ Describe a numbering plan.

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Chapter

1

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Evolution is the process of something changing over time into a more complex state where it can better adapt to its environment. Evolution typically is triggered only when

outside forces require changes to be made. Technology also evolves into newer and more useful tools over time. While the analog phone is still around, advances have been made and telephones have evolved into fully digital devices. Even more recently, we’ve seen more and more voice running over IP networks that share the same cabling and routing functions with data networks.

But throughout this telephone evolution process, many of the traditional interfaces, signaling protocols, and setups remain unchanged. In order to understand voice networks of today, we must fi rst take a step back in time to discuss traditional telephony topics. Once you have a solid foundation, you can see how many of these elements have either remained the same or evolved over time to improve voice networks as they transition from circuit-switched networks to packet-switched networks.

Chapter 1 will start off covering traditional telephony devices. This includes legacy analog and digital phones as well as a look at components within public telephone networks. We will then move on to the two private telephone network types in most organizations. Lastly, this chapter will cover Cisco’s take on IP telephony networks and how it breaks down components into separate functionality categories and deployment models.

Understanding Traditional Telephony ComponentsIn 1875, Alexander Graham Bell invented the telephone, a device that transmits and receives sound, most commonly human speech. The telephone houses a microphone that callers speak into. With a standard analog telephone, the speech is then transported across a pair of copper wires in the form of an electrical signal.

As the popularity of telephones grew, companies began providing a telephone network that was used to interconnect multiple phones throughout a region. Today, public telephone networks are a mixture of analog and digital circuits and trunks that interconnect and cover the globe.

Telephone systems can be split into public and private sections. The private side consists of equipment owned and maintained by an individual user or business. The public side is owned and maintained by the telephone company, and this service is paid for by the

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Understanding Traditional Telephony Components 3

individual or business owner who wants to use public phone services. The public switched telephone network (PSTN) is the network that interconnects telephones found in homes and businesses throughout towns, cities, countries around the world. It used to be that the PSTN consisted solely of analog circuits. The fi rst analog circuit was just two wires, and it was responsible for carrying a single telephone call. As technology improved, both the public- and private-side equipment became more sophisticated. Private businesses could own and maintain their own phone switches. These phone switches could then be interconnected by trunk lines that were specifi cally designed for the transport of voice services between phone switches. In this fi rst section, we will investigate the traditional telephony components that make up the private and public telephone network.

Telephony Edge Devices

The edge is the part of the phone system that end users interact with to make and receive calls in their purest form. Traditional telephony edge devices can be divided into two categories: analog and digital telephones. But even traditional telephony devices have evolved to include more advanced features to make the calling experience a better one. Here is a closer look at each of these phone types.

Analog Telephones

Analog edge devices are still somewhat common in homes and small business environments. The analog telephone is commonly directly connected to the PSTN, so all of the backend intelligence is the responsibility of the service provider, and the phone user is simply responsible for purchasing and maintaining their analog telephone, which is a very simple device. Some businesses still use analog PBX (private branch exchange) systems, although they are becoming rare. Connecting an analog phone to a PBX provides additional capabilities to the phone such as voicemail with message-waiting indicators, call hold, and personalized ringtones. Other than that, the features of analog telephones are very limited.

Digital Telephones

Digital telephone devices use special hardware to convert analog voice streams into a digital data stream. Most legacy PBX systems are digital. It is also important to note that the digital handsets of most of these digital PBX systems are proprietary. It is rare to be able to mix and match different digital phones within a single digital PBX.

Phone Switches

On the public side of the overall telephony, there are public phone switches and private phone switches. A PBX or key system can be installed by a private party to provide a multitude of private telephone services to phones located within this private network. The differences between a PBX and a key system are detailed later in this chapter. Extension-to-extension

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4 Chapter 1 ■ An Introduction to Traditional Telephony and Communications

dialing, multiple lines, voicemail, call waiting, and call forwarding are just a few of the services that private switches can provide.

A private switch does nothing when a call needs to be placed to a phone that is located outside the private network. This is where a connection to the PSTN comes into play. Privately owned phones and/or private switches connect to public telephone switches. These switches handle public call routing and signaling.

The Central Office

A PSTN central offi ce is the fi rst major stop where a public telephone line terminates. Central offi ce (CO) is a term used to describe a geographically located offi ce that houses PSTN switch equipment. Home and business lines are run back to the central offi ce and connect to the PSTN switches. The CO has large trunk lines that further multiplex the phone lines and interconnect this central offi ce with the larger national and global telephone network.

In spite of the name, most modern central offi ces are not really offi ces at all but underground bunkers of sorts. The switches and cabling are built underground for two main reasons:

� To help protect the cabling and switch equipment from lightning strikes

� To limit the amount of electromagnetic radiation emitted by the lines and equipment, which can interfere with analog radio and over-the-air television signals

To better understand where the CO fi ts into the PSTN, imagine that you are at your offi ce and need to call a customer of yours who is right down the street. Their telephone number is 555-1717. If you are connected to the same CO (which is likely), then your call would be directed out of your offi ce on the PSTN line and reach the central offi ce switch equipment. That switch equipment would then look up the destination number of your customer and discover that the destination terminates within the CO. The switch would then use telephony signaling protocols to complete the connection and ring your customer’s phone, as shown in Figure 1.1.

F I GU R E 1.1 A PSTN call within the same central office

PSTN

central

office

555-1717

555-1717

Called party

Calling party

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CO switches can also be compared to IP routers in a sense. From an IP router perspective, packets enter a router interface, and they contain an IP address that identifi es the destination device. The router uses the IP address to perform a routing table lookup to see which router interface is the shortest path to the destination. A CO switch is similar in that it too contains a table. But instead of IP addresses, the table consists of telephone digits. These digits have a hierarchical structure similar to IP addresses. A hierarchical structure helps to reduce the lookup table size and makes decision making faster and more effi cient. When calls enter a switch, the destination number is effi ciently matched within the CO switch lookup table.

The Local Loop

The local loop is the physical connection that connects a customer’s private telephone equipment to the PSTN central offi ce. The loop is typically copper wiring and carries single phone lines or multiple lines in the form of T1/E1 connections.

The local loop is sometimes referred to as the “outside plant” in very large businesses with multiple connections to the CO.

Figure 1.2 shows an example of a business that has its private telephone equipment connecting to the PSTN CO through the local loop wiring.

F I GU R E 1. 2 A local loop

PSTN

Central

office

Local loop

PBX PSTN

demarc

The customer’s site has a termination point called the demarcation point (demarc). This point separates the customers’ house wiring from the PSTN’s wiring and assigns the physical cabling responsibilities accordingly. If a problem occurs on the PSTN lines, the PSTN may visit the customer’s site and will troubleshoot up to the demarc. If the problem is on the customer’s side of the demarc, it is the responsibility of the customer to fi x.

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Trunks

Traditional telephony trunks are circuits that interconnect voice switches. There are three distinct types of trunk lines:

� Tie trunks

� Central offi ce trunks

� Interoffi ce trunks

The trunks themselves are similar for the most part except for the types of phone switches (either public or private) they interconnect with. The following sections describe each telephony trunk type in more detail.

Tie Trunks

A tie trunk (or tie line) is a dedicated voice circuit that directly connects two PBX switches. This point-to-point connection is commonly used within private organizations to tie multiple telephone systems together, as shown in Figure 1.3.

Calling

party

Called

party

4104

PBX-A

3XXX

PBX-B

4XXX

Tie trunk

4104

4104

F I GU R E 1. 3 A tie trunk

So why would a business ever need to have more than one PBX? There many reasons, but these are some of the more popular ones:

� Migrating from one PBX system to another

� A merger of two or more businesses resulting in the need to combine PBX systems

� A business or organization with multiple voice management groups that control their own independent PBX systems

As a CVOICE candidate, you probably have an IP networking background, so these reasons can be best compared to the migration and merging of IP networks. For example, a merger between two separate PBX systems is similar to a merger of two separate IP networks. The networks may not use the same routing protocols and therefore must either be reconfi gured so they use the same routing protocol or confi gured to redistribute into one another. At a very high level, the same challenges found in migrating two IP systems are

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similar to merging two PBX systems. In both situations, similar planning methodologies are required to successfully merge the two systems.

Central Office Trunks

Central offi ce trunks are the circuits that connect a private business PBX to the PSTN. When organizations have large PBX systems, having many users increases the number of simultaneous calls, which requires multiple outside lines to the PSTN. The most effi cient and economical method is to have a trunk connection from the private PBX to the local PSTN CO switch, as shown in Figure 1.4.

PSTN

Central

office

Calling party

External calls

External calls

CO trunk

PBX

F I GU R E 1. 4 A CO trunk

Again, keep in mind that the physical wiring between the private PBX and the CO is known as the local loop. In a large-business scenario, the local loop can be also referred to as the central offi ce trunk.

Interoffice Trunks

Interoffi ce trunks are the backhauled connections that interconnect central offi ces. Central offi ces that are connected with interoffi ce trunk lines are considered to be interexchange connections. It’s easiest to understand interoffi ce trunks in terms of local vs. long-distance charges; a call whose routing goes no higher than an interoffi ce trunk is considered local. For example, imagine you are in your offi ce and need to call someone on the other side of the city with the number 555-1717. You pick up the phone and dial the 7-digit (or sometimes 10-digit) number. The dialed digits (known as DTMF, or dual-tone multi-frequency, as discussed in Chapter 2, “Understanding Analog and Digital Voice”) are interpreted by your local CO telephone switch, which determines that the destination phone does not reside within the local CO but at a CO that is accessible through an interoffi ce trunk connection. The phone switch seizes one of the lines on the interoffi ce trunk and communicates with the neighboring CO switch to help terminate the call at the correct phone across the city. Figure 1.5 depicts the call process fl ow using an interoffi ce trunk between COs.

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This type of interoffi ce trunk would likely be considered a local call instead of a long-distance call because the call uses the interoffi ce trunk line to complete the call as opposed to moving farther up the PSTN hierarchy. In North America, it used to be that local calls were strictly defi ned by the area code they belong in. Only 7-digit dialing was considered to be a local call and therefore did not incur long-distance charges. Over time, and due to the U.S. government stepping in and breaking up the AT&T monopoly, it became obvious that the area code method for determining local vs. long distance would not be able to function in the future, for two reasons:

� The deregulation of AT&T by the U.S. government meant that the FCC must decide what would and would not be considered a long-distance call. The FCC came up with the concept of Local Access and Transport Areas (LATAs). These LATAs were supposed to be used by the newly formed “Baby Bell” companies to determine what was considered long distance and what was not. LATAs were broken up mainly by population and oftentimes overlapped state lines. LATAs oftentimes broke cities and towns into multiple zones that would have necessitated the need for a long-distance call that may have been right across the street.

� The rapid growth of telephone number usage in large cities required multiple area codes to overlap in a single geographical area. It became possible that a telephone in one offi ce might have a different area code than a different telephone in the same building. No longer did area codes actually mean a different geographical area as they were fi rst intended.

Because of the LATA and overlapping number confusion found in the U.S. telephone numbering system, you will fi nd that certain 10-digit dialing is now considered to be local.

National and International Calling PSTN

For the bigger picture, we need to distinguish between, local, long distance, and international long distance. This network setup varies from country to country, but at its core, there is a three-step PSTN hierarchy, as shown in Figure 1.6.

Calling

party

Called

party

555-1717

PBX

4XXX

Interoffice

trunk

55

5

-1717

555-1717

PSTN

Central

office

PBX

4XXX

PSTN

Central

office

F I GU R E 1.5 An Interoffice trunk connection

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Understanding Private Telephony Phone Systems 9

Telephone calls between COs that have interoffi ce trunks are considered to be local calling. If the telephones are on different networks that are not interconnected using interoffi ce trunks but fall within some type of border (such as by phone company, state, or nation), the call is considered to be at the next level of the PSTN hierarchy, called the interexchange network. Typically, long-distance charges begin to apply. Lastly, if the call is placed between two international borders, it is considered to fall within the highest level of the PSTN hierarchy, called the international network. International long-distance charges begin to apply at this level.

Understanding Private Telephony Phone SystemsIn a business environment with multiple employees, you will quickly see that having individual telephone lines run in from the PSTN is not the most effi cient or economical method for providing voice services. It would be extremely rare to fi nd a time when every employee needed to access the PSTN at once. In fact, calculations show that the telephone-to-PSTN line ratio is quite low. Therefore, business environments often implement some sort of intelligence that allows multiple employees to have their own telephone handsets while sharing PSTN lines. The two traditional telephone systems available are the key system and the PBX. The next two sections briefl y explain the differences between these two systems.

Central office (local

calling)

Interexchange network

(long distance calling)

International network

(international long

distance)

Lo

ca

l to

In

tern

ati

on

al ca

ll fl

ow

F I GU R E 1.6 The PSTN local-to-international hierarchy

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Key System

Very small businesses may choose to implement a key system, which is a simpler solution and easier to manage than a PBX but offers fewer features. All of the telephone handsets in a system are identical, and each phone shares the same small group of PSTN external numbers, indicated by lights and selected by pressing buttons. Key systems do not assign unique telephone numbers to individual phones. This ensures that anyone in the offi ce can answer an incoming call to any line. The key system is often called a shared-line system, because of how lines are confi gured on the phones.

PBX

A private branch exchange (PBX) system is similar to PSTN switches owned and operated by the PSTN. In fact, many PBX systems found in very large organizations use identical switching equipment. Traditionally, businesses with employees of 20 or more will choose to implement a PBX because of the more advanced features available to end users and the scalability to grow both internally by adding additional handsets and externally by connecting to other PBX systems using tie trunks or to the PSTN using CO trunks. While PBX systems can either be analog or digital in nature, most legacy PBX systems in use today use a digital transport method.

One of the major usage differences between PBX and key systems is where the calls that originate within them are going. With key systems, because the businesses are typically small, very few internal-to-internal calls are made. By contrast, a large percentage of calls made on a PBX system are employee-to-employee calls. That is why it is very common for PBX systems to use truncated numbers for internal dialing. These truncated numbers are called extensions and are typically three to fi ve digits in length.

When Is It Time to Upgrade a Key System?

Kevin began his language service business back in the spring of 1998. When his business was just starting up, his only employees were himself and four other persons, who each took a portion of the sales and accounting duties. Kevin decided at that time to implement a key system. To Kevin, this was a logical choice because he needed only three phone lines for all his calls. Each employee had all three telephone numbers confi gured on their phone. When a call came inbound from the PSTN (which was rare), any one of the employees could answer the line. If the caller needed to speak to someone directly, it was a simple process of placing the call on hold and yelling to the other side of the small offi ce for the proper person to pick up the line.

Over the years, Kevin’s business grew and with it the offi ce space and number of employees. The key system continued to be suffi cient until an interesting phenomenon

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Understanding the Unified Communications Model 11

Understanding the Unified Communications ModelThe Cisco CVOICE exam requires that students have a basic understanding of the end-to-end Cisco components involved in Internet Protocol Telephony (IPT), which is a method used to transport voice communications over an IP network, and Voice over IP. Cisco groups its components into four categories:

� Endpoints

� Applications

� Call processing agents

� Network infrastructure

Endpoints

Cisco has a plethora of both hardware- and software-based IP phones for nearly any voice situation. In Cisco’s Unifi ed Communications model, an endpoint can be any end device or software that interacts with Unifi ed Communications hardware and/or software. This is because Cisco lumps together multiple communications methods, including voice, videoconferencing, and instant messaging, under one umbrella of services. The following section is meant to give a broad overview of many of the different product features. It is not a complete list of all of the phones available, however. These models are also continuously being updated. While Cisco does not make analog telephones, it does offer hardware

occurred. After a while, the inbound calls that were directed to a specifi c employee increased signifi cantly. This is because as the business grew, employees became specialists in a specifi c part of the organization. No longer could any employee handle any request. In addition, as the number of employees increased, a need arose to have individual voicemail boxes, which a key system cannot handle, because no employee has a dial-in number or extension.

Because of this, Kevin had to migrate to a new Cisco CUCM Express solution, which he implemented as a PBX switching system. Now Kevin has the PBX set up so each one of his 22 employees has their own unique telephone extension to make and receive both internal and external calls. The migration from a key system to a PBX system let Kevin’s business better adapt to growth.

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solutions that convert analog into IP for use on modern networks. This section examines the following categories of IP phone endpoints:

� Wired IP phones

� Wireless IP phones

� Software IP phones

� Videoconferencing phones

� Analog-to-IP adapters

This should give you an understanding of the wide spectrum of Unifi ed Communications devices available for implementation.

Wired IP Phones

When most people think of IPT, the fi rst type of IP phone endpoints that they think of are standard wired IP phones. This is likely because they most closely resemble analog telephones of old. However, newer IP phones are beginning to look more like computers than telephones, with the LCD displays, soft keys, and user-programmable features. Cisco divides its wired IP phone systems into two major categories: small-business and enterprise-class phones.

Wired IP Phones for Small Business

Cisco’s small-business IP phones include the SPA 300 and 500 series. These entry-level phones are designed to work only with the Cisco Communications Manager Express call agent solution. They are part of the Cisco Smart Business Communications System (SBCS) suite of products and fully interoperate with products such as the UC500 series CUCM Express and Unity Express voicemail products. This is because the phones specifi cally support the Cisco proprietary Smart Phone Control Protocol (SPCP), which only the UC500 series platform call agents support. In addition, the 300 and 500 series phones also support the Session Initiation Protocol (SIP) for call signaling on an IP network for connecting to an Internet Telephony Service Provider (ITSP). An ITSP is a fairly new service in which small businesses can pay a monthly fee to have a service provider manage the backend IPT hardware while enjoying the added features and cost savings that IP phones provide over PSTN solutions. All that is needed is a high-speed Internet connection to the ITSP. Calls are routed to the ITSP across the Internet, and the ITSP then routes calls out to the PSTN on its end.

Wired IP Phones for the Enterprise

The majority of Cisco IP phone offerings can be found in the enterprise class of hardware. These X900 phones are further categorized by a numbering system in which X is the number of a specifi c series. Within these categories, there are minor differences between the features the individual phones offer. As of the writing of this book, the enterprise-class IP phones include the following:

� 9900 series

� 8900 series

� 7900 series

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� 6900 series

� 3900 series

Each of these phone series is designed to meet a specifi c market niche. For example, the 3900 series phones offer basic functionality and do not include many of the add-on bells and whistles that some of the high-end models tout. Cisco markets this line of phones for use in lobbies, manufacturing fl oors, and retail-outlet fl oors where a basic-use phone may be needed by employees and people in public-access areas.

It is not necessary to know all the wired/wireless IP phones that Cisco offers. There simply are too many to list here, so we chose to discuss only the most unique. If you want to investigate all of Cisco’s IP phone offerings, you can refer to Cisco’s IP phone product web page at http://www.cisco.com/en/US/products/sw/voicesw/index.html#~all-prod and begin your research under the “IP Communications” section.

Wireless IP Phones

The Cisco 7900 series of phones offers the majority of different models. This includes the two models of wireless IP phones currently:

7921G Wireless IP Phone This phone can operate on 802.11a/b/g networks.

7925G and 7925G-EX Wireless IP Phones These phones can operate on 802.11a/b/g networks. In addition, they offer Bluetooth 2.0 support and a hermetically sealed and rug-gedized case for heavy-use situations.

Unified Communications IP Soft Phones

Cisco has several software-based IP phones that let users make and receive voice calls on computer hardware. The requirements are a compatible PC, a microphone, and speakers/headphones. Once one of the following applications is installed and connected to a compatible version of Cisco Unifi ed Communications Manager (CUCM) server, you can make and receive phone calls without an actual telephone handset. Here are the three primary Cisco software IP phones available:

Cisco IP Communicator This software-based IP phone behaves just like a 7970 hardware-based phone. That means that everything a hardware phone can do, the IP Communicator can do as well. The application can be installed on Microsoft Windows XP, Vista, and Windows 7 operating systems.

Cisco Unified Personal Communicator This software application integrates, voice, voicemail, instant messaging, and other features into a single application that can be installed on the latest Microsoft Windows and Mac OS–compatible computers.

Cisco Unified Mobile Communicator With the increased mobility of business phone users, thanks to 3G and 4G availability, Cisco created the Unifi ed Mobile Communicator software package to run on popular smartphones such as Apple’s iPhone and the RIM Blackberry. The software lets users interact with the Cisco Unifi ed Communications

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platform remotely to accomplish tasks such as retrieve missed calls, join MeetingPlace conferences, and even make and receive calls on a mobile phone, giving the impression to the called/calling party that you are making the call from your desk phone extension. Not only does the Mobile Communicator software make life easier on the mobile user, but it can also dramatically decrease mobile phone bills by limiting roaming charges, because calls made and received through the software run through the offi ce telephony infrastructure over the phone’s wireless data connection.

Video Phones and Tablets

Some hardware- and software-based phones integrate voice and video. The four primary Cisco endpoints in this category are as follows:

7985G IP Video Phone This phone features a large, color LCD display and built-in high-resolution camera for videoconferencing to other 7985G phones as well as all of Cisco’s other video hardware and software applications.

9951 IP Video Phone This phone features a touchscreen color display, built-in high-resolution camera for videoconferencing and collaboration applications, and Ethernet or Wi-Fi connectivity.

Cisco Video Advantage This products works with the Cisco VT Camera II USB hardware to make and receive videoconference calls on a desktop PC running Microsoft Windows software. The VT Camera II plugs into a PC through a USB port. Users make and receive the video calls using their Cisco Unifi ed IP desktop phones or the Cisco IP Communicator software installed on the same PC as the Cisco Video Advantage software.

Cisco Cius Tablet The Cius is a new product from Cisco that combines Unifi ed Communications voice and video functionality with additional PC functionality. The tablet can connect to a Cisco Unifi ed Communications system, either wired or wireless, from inside the enterprise. An optional 3G/4G wireless option is available to use the tablet as a mobile communications tool from outside the offi ce. Figure 1.7 shows the Cisco Cius tablet.

F I GU R E 1.7 The Cisco Cius tablet device

Courtesy of Cisco Systems, Inc. Unauthorized use not permitted.

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Analog-to-IP Adapters

Some people just can’t let go of their analog endpoints for one reason or another. Much of it has to do with the high cost of replacing all phones within a system. Another important reason is that many businesses still rely on analog fax machines for their day-to-day business operations. Cisco has anticipated this and has two major analog-to-IP adapters to get these incompatible systems to interact on an IP phone network. Analog telephony adapters (ATAs) are appliances that have an Ethernet port to connect to an IP LAN. They then have two or more RJ-11 ports for connecting to analog telephones. The appliances then use software to convert the analog signal into a digital IP packet for proper transport on any IP network. The two primary solutions available from Cisco are these:

ATA 180 Series These are small point-solution appliances for connecting two analog desk phones, conference phones, or fax machines to an IP network. These devices are good for small businesses or anywhere only two analog phones are needed in one geographic location.

VG200 Series The VG200 series appliances are Cisco’s newest analog-to-IP devices that offer additional integration to the Unifi ed Communications features available. The form factors of these stand-alone analog gateways include the ability to connect 2, 4, 24, or even 48 analog devices to a single appliance. The two- and four-port models are scheduled to completely replace the ATA 180 series once the 180 series goes end-of-life. The VG224 and 248 are high-density appliances that run on special IOS software that runs on ISR (Integrated Services Router) equipment.

Applications

In addition to the Unifi ed Communications platform calling features, Cisco provides value-added Unifi ed Communications applications that seamlessly integrate into the product lineup. These applications include voicemail functionality with Cisco’s Unity lineup, Emergency Responder for 911 services, conference call applications in the form of the Cisco Conference Connection suite, and billing applications. These add-on telephony applications reside on dedicated hardware and software platforms and bolt into the Unifi ed Communications call processing agents that are described next.

Call Processing Agents

Call processing agents are the brains behind IP call-processing and call-control mechanisms on a LAN. From a Cisco prospective, call agents are Cisco Unifi ed Communications Manager solutions. These were previously called Cisco Call Managers. Our discussion of call agents will look at the three different Cisco call agent call-control responsibilities:

� Call agents are responsible for the setup and teardown of telephone endpoints on the local network.

� Call control agents are used for IP telephone endpoint registration to the call agent.

� Voice gateways are used to bridge voice networks.

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Cisco Call Agent Solutions

Cisco offers three primary call agents to handle call processing for small, medium, and large organizations. It is important to know the primary differences between the three solutions. In addition, the 642–437 exam requires that you be able to confi gure basic settings on the Unifi ed Communications Manager Express. This material is covered in Chapter 8, “Confi guring and Managing the CUCM Express,” of this study guide. Here’s a brief look at the three call agent solutions:

Cisco Unified Communications Manager Cisco Unifi ed Communications Manager (CUCM) is Cisco’s fl agship call agent. It is a hardware appliance that runs on a hardened Linux operating system. The current CUCM version is 8.0, which includes a number of feature enhancements over versions 6 and 7. Each server appliance is capable of handling up to 7,500 endpoints and can be clustered to support up to 30,000 endpoints.

Cisco Unified Communications Manager Business Edition Cisco Unifi ed Communications Manager Business Edition (CUCMBE), Cisco’s medium-sized solution, is basically the full-blown CUCM solution except for some key differences. The fi rst is a limit of 500 endpoints on each appliance. It also does not offer the high-availability redundancy features found in the CUCM. One major benefi t of the CUCMBE is that it offers an integrated voicemail system, called Unity Connection, which runs on the same hardware as the call agent software. This helps lower customer costs by allowing one piece of hardware to be used for both purposes.

Cisco Unified Communications Manager Express The Cisco Unifi ed Communications Manager Express (CUCME) call agent differs greatly from the CUCM and CUCMBE in the fact that the express software runs on Cisco routers. That means that the CUCME runs a specialized version of the Cisco IOS. In addition, specialized cards or interfaces can be installed into Cisco routers for voicemail access using Unity Express software. This solution lets small businesses have a fully functional IP data, voice, and voicemail solution contained in a single appliance. The CUCM Express is geared to small businesses with up to 250 endpoint devices.

Cisco Call Control Agent Solutions

Call agents are responsible for handling IP phone endpoint setups so that the phones receive extension numbers and other calling features unique to each phone. The IP phone endpoints register to the call agent and communicate with it each time a call is placed on the network. These are all functions of call control. When a user picks up an IP phone that is registered to a call agent, that phone relies on the call agent for things such as dial tone and other supervisory and informational signals (discussed in Chapter 2). When the end user dials in a telephone number, the address-signaling information is sent from the phone to the call agent. The call processing agent then has various settings and rules in place to either permit the phone to call this number or deny it. For example, if the end user attempts to call an international number on their desk phone, the call agent may deny this request so the business does not incur expensive long-distance charges. If the call is allowed, the call

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agent performs signaling between the source IP phone and the voice gateway, as shown in Figure 1.8.

M

Off-network

phoneIP phone

Call agent

PSTN

Signaling Signaling

V

F I GU R E 1. 8 Call setup signaling through the call agent

From a Cisco hardware perspective, call agents and Cisco IP endpoints can communicate call setup information using either the Cisco proprietary Skinny Client Control Protocol (SCCP) or the IETF-defi ned Session Initiation Protocol (SIP) method. By default, Cisco call agents and most Cisco IP phones are confi gured for SCCP signaling. Signaling between the call agent and the voice gateway (as shown in Figure 1.8) can be H.323, SIP, MGCP (Media Gateway Control Protocol), or SCCP.

Once the call signaling is established between the source and destination phones, the voice stream is transported directly between the phone and the voice gateway, which is the fi nal hop on the IP network before it must be converted for proper transport on the PSTN. The transport of voice on an IP network uses a separate protocol, as shown in Figure 1.9. By using a separate and direct protocol for voice transport, voice packets are sent on the most direct and effi cient path.

M

Off-network

phoneIP phone

Call agent

IP voice packet transport

Signaling Signaling

PSTNV

F I GU R E 1. 9 Voice transport

The protocol used to transport voice is the Real-Time Protocol (RTP) and is discussed in detail in Chapter 4, “The VoIP Path-Selection Process,” of this book. When the phone conversation is fi nished, both phones will again communicate call control information with the call agent to end the call, just as they did with the setup signaling information shown in Figure 1.8.

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Cisco Voice Gateway Solutions

One major topic of the CVOICE exam is the functions of voice gateways on an IP network and how to confi gure them. Voice gateways are a critical component of an IP network for several reasons. Primarily, a voice gateway sits on the border between an internal IP voice network and the public switched telephone network or a legacy PBX. Because these two networks are incompatible with each other, it is the responsibility of the voice gateway to translate between them. In order to accomplish this task, the voice gateway uses various hardware ports to connect to the PSTN. It uses special hardware called digital signal processors (DSPs) to translate from one medium to another for proper interoperation. In addition, voice gateways are confi gured to speak one or more signaling protocols that are used to properly route calls to and from the IP call agent, which may be one of the Cisco Unifi ed Communications solutions described earlier in this chapter. These signaling protocols are as follows:

� H.323

� SIP

� MGCP

� SCCP

Each of these protocols has benefi ts and drawbacks. Careful consideration must be made during the voice network design phase to choose the signaling protocol that best fi ts the following:

� Call agent hardware and software

� Voice gateway hardware and software

� Legacy hardware that needs to interact with the IP solution

DSPs are also used to facilitate other responsibilities (such as offl oading call conferencing duties) for the proper operation of voice on an IP network as well as for value-added features. All of these topics will be covered in depth in Chapters 5, “VoIP Design Options,” and 6, “Confi guring Voice Gateway Ports and DSPs.”

Voice Gateway Hardware Components

Cisco voice gateways are routers that have special IOS software designed to support voice interface cards and voice signaling protocols. Specifi cally, voice gateway IOS software operates on some older router platforms, including these:

� 1700 series

� 2600XM series

� 3700 series

In addition, voice gateway IOS runs on newer Cisco Integrated Services Routers (ISRs). These include the following router platforms:

� 1800 ISR series

� 2800 ISR series

� 3800 ISR series

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The ISR router series platform is slated to become end-of-life soon and will be replaced by the ISR G2 (Generation 2) series platform as follows:

� 2900 ISR G2 series

� 3900 ISR G2 series

While there is a 1900 ISR G2 platform to replace the 1800 series, voice services will not be available. Table 1.1 compares the 2900 and 3900 ISR G2 voice-capable routers and the Unifi ed Communications voice capabilities they offer.

TA B LE 1.1 Comparison of the 2900 and 3900 series ISR G2 platforms

UC Feature 2900 Series ISR 3900 Series ISR

Conference call support Yes Yes

DSP support PVDM 2/3 PVDM 2/3

Max SRST calls 250 1500

Max SIP sessions 600 2500

Max digital voice calls 400 660

Max FXO ports 40 60

Max BRI ports 24 38

Lastly, voice services can be integrated into large enterprise and service provider hardware, including these series:

� 1000 ASR series

� 9000 ASR series

� 6500 series

� 7200 series

� 7600 series

� 12000 series

� AS5400 series

� AS5800 series

The actual implementation of voice on the hardware listed here is different from the ISR platform and is outside the scope of this study guide.

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The primary focus of the CVOICE exam is on the ISR series of router equip-ment. It is important to understand which equipment is considered to be part of the ISR lineup.

Network Infrastructure

The fi nal Unifi ed Communications model component in Cisco’s design is the IP network itself. This consists of standard IP devices, such as routers, layer 2/3 switches, and fi rewalls, that transport both regular IP data and Unifi ed Communications traffi c over the same physical network. The major point to note in a Unifi ed network infrastructure is the importance of Quality of Service (QoS) techniques that must be understood and properly deployed to ensure proper transport of time-sensitive traffi c such as voice and video.

Unified Communications Deployment ModelsCisco presents four different deployment models that it recommends for use within a production network for the entire Unifi ed Communications version 8.X suite. These models have remained fairly static over the years, but the terminology has changed to refl ect the fact that Unifi ed Communications offers more than just voice services in today’s networks. Although this section discusses placement of the call processing agents, in reality other UC servers and services can be centralized or distributed to perform the same way for the services they provide. Each deployment has its benefi ts and potential drawbacks. The primary differences between the four models discussed next depend mostly on the following characteristics:

� Number of users supported

� Physical location of users

� Amount of WAN bandwidth and QoS controls

� Ability to offer alternative methods to achieve high availability (HA)

The Centralized Services Deployment Model

The centralized services deployment model is ideal when a single building or a group of buildings in a campus is interconnected in a LAN environment. A single call processing agent can be used, or multiple call processing agents can be clustered to provide scalability and high availability to voice services. The benefi ts of this model derive from having a single administration point with which to manage the Unifi ed Communications (UC) services. A major drawback is the lack of scalability if your UC needs extend outside the single location site. Figure 1.10 shows a typical centralized services deployment model.

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Unified Communications Deployment Models 21

Campus

M PSTNV

F I GU R E 1.10 The centralized services deployment model

The Distributed Services Deployment Model

The distributed services model is useful when you have a large campus site and a handful of smaller remote sites that are connected to the primary site using high-speed WAN connections. This model often represents the classic hub-and-spoke look. While this is a great model in many instances, you should plan for high availability in case of a WAN outage. Cisco commonly suggests using either Survivable Remote Site Telephony (SRST) or a CUCME in SRST fallback mode. Both of these features allow the remote sites to route calls to outbound PSTN links in the event of a WAN failure, at which point the remote site cannot access the call processing agent. Figure 1.11 shows the distributed services deployment model.

Campus

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VPSTN

Branch Branch

SRST

V

IP WAN

V

V

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F I GU R E 1.11 The distributed services deployment model

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The Inter-Networking of Services Deployment Model

The inter-networking of services model is used for organizations with multiple large and geographically dispersed locations. In this situation, call processing agents are distributed and located at the various sites and act largely as independent single-site deployments. Local calling therefore never traverses over WAN links, which conserves bandwidth. This model is also useful when WAN links are not reliable or do not have enough bandwidth for handling voice traffi c in addition to data traffi c. Calling between sites can be sent either across an IP WAN or through the PSTN. Figure 1.12 shows this model, also called multisite with distributed call processing.

F I GU R E 1.12 The inter-networking of services model

Campus

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IP WAN PSTN

M

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Campus

M

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Campus

V

V

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The Geographical Diversity Deployment Model

The fi nal deployment model is the geographical diversity deployment model. In this model, the organization again has multiple geographically dispersed locations with a large number of users. The call processing agents are distributed as in the case of the inter-networking of

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Summary 23

services model. The difference this time is that the geographical diversity call-processing agents work as a single cluster across interconnecting IP WAN links. As with the distributed services model, it is extremely important to have WAN connections with ample bandwidth and QoS enabled for voice. One benefi t in this model is that local site calling is contained within the LAN and does not traverse the WAN. In addition, WAN links in a mesh design can provide some form of redundancy to the point where SRST is not required. Figure 1.13 shows this model, also called clustering over IP WAN.

Campus

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IP WAN PSTN

M

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Campus

M

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Campus

Cluster

V

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F I GU R E 1.13 The geographical diversity model

SummaryYou should now have a solid understanding of the hardware and software components involved in traditional telephony systems. In addition, this chapter covered the two different types of private telephone equipment and when a business might choose to implement a key system or a PBX based on internal vs. external calling patterns. Lastly, we covered Cisco’s IPT component and deployment models.

We’ll cover many of these same topics again throughout this book in much more detail. This chapter was written to give you a 30,000-foot view of traditional telephony so that

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when we cover newer and more advanced topics on voice in future chapters, you’ll be able to understand how the voice network has evolved to the point where it is today. Many things have changed while some things remain the same.

Exam EssentialsKnow the two different types of traditional telephony edge devices. The two main types of edge devices are analog and digital phones. Besides the obvious difference of handling voice services as analog or digital formats, digital telephones often offer additional service features and are more commonly found in legacy business PBX systems.

Know what a phone switch is. Phone switches are responsible for routing calls throughout a voice network. They can be privately owned as is the case with a PBX, or they can be part of the PSTN.

Know what a central office is and where it is located in relation to a business. The central offi ce is the fi rst telephone switch that personal and business telephones reach on the PSTN. The physical cabling between the privately owned telephone equipment and the CO is called the local loop.

Understand the difference between tie, central office, and interoffice trunks. Tie trunks connect two PBX systems. Central offi ce trunks connect a PBX to the CO. Interoffi ce trunks interconnect two COs.

Understand the three-tiered PSTN hierarchy. The PSTN routes calls based on central offi ce, interexchange, and international networks. The PSTN uses a hierarchical network structure where local calls are routed through the central offi ce, national calls through the central offi ce, and interexchange and international calls through all three tiers—the central offi ce, interexchange, and international networks.

Understand the difference between a key system and a PBX. Key systems are used in smaller environments, have few features, and do not have unique extensions. PBX systems are found in large businesses, have many features, and pool external numbers while having unique internal extension numbers.

Know the four Unified Communications model tiers and which products fall within each tier. The endpoints tier contains hardware and software the end user interacts with. The applications tier contains various value-added applications used in the UC lineup. The call-processing layer is the brains where call processing takes place. This is where the UC Manager resides. Lastly, the network infrastructure is the IP network equipment that transports voice and data as well as employs QoS.

Understand the four Unified Communications deployment models. Know the four models and when they should be implemented. Understand that choosing one model over another depends on several factors, including number of users, physical location of users, amount of bandwidth, and alternative methods to achieve HA.

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Written Lab 1.1 25

Written Lab 1.1Write the answers to the following questions:

1. These PSTN sites house telephone switch equipment that directly connects to personal telephones and/or offi ce PBX switches.

2. In large environments, the local loop is also called the outside .

3. Name the three types of voice trunks in a traditional telephony network.

4. What are the four categories of the Cisco Unifi ed Communications model?

5. What is a private phone system that uses shared-line extensions?

6. The Cisco IP Communicator software resides in which UC model category?

7. What hardware interconnects two incompatible voice networks?

8. A CUCM is also called a agent.

9. What hardware device allows a user to connect analog telephone devices to an IP network?

10. What UC deployment model is useful when you have a large campus site and a handful of smaller remote sites that are connected to the primary site using high-speed WAN connections?

(The answers to Written Lab 1.1 can be found following the answers to the review questions for this chapter.)

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Review Questions

1. What is the name of the first major stop that a public telephone line makes when leaving the customer site?

A. Tie trunk

B. CO trunk

C. Local loop

D. Central office

2. There is a specific point within a customer’s site that defines the physical cabling responsibilities between the private owner and the telephone company. What is this point referred to as?

A. Demarc

B. Inside wiring

C. Outside wiring

D. Local loop

E. House wiring

3. What type of voice trunk directly connects two PBX systems?

A. Demarc

B. Tie trunk

C. CO trunk

D. Interoffice trunk

4. What type of voice trunk directly connects two central offices?

A. CO trunk

B. Interoffice trunk

C. Tie line

D. Interexchange trunk

5. Central offices maintain pools of what numbers?

A. Subscriber code

B. E.164 code

C. Area code

D. Interexchange code

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Review Questions 27

6. At what point of the PSTN hierarchy described in this Study Guide will a caller begin incurring long-distance charges?

A. Central office

B. Local loop

C. International network

D. Interexchange network

7. Which Cisco IP phone does not support Cisco’s proprietary SCCP signaling protocol?

A. 7925G series wireless phone

B. SPA 500 series phone

C. Cisco IP Communicator softphone

D. All Cisco phones support SCCP

8. The Cisco Emergency Responder belongs in what UC model category?

A. Network infrastructure

B. Applications

C. Call processing agents

D. Endpoints

9. What is an analog-to-IP adapter used for?

A. To translate between analog and IP signaling protocols for proper transport on the PSTN

B. To translate between analog and IP signaling protocols for proper transport on an IP network

C. To translate between voice and data signaling protocols for proper transport on the PSTN

D. To translate between voice and data signaling protocols for proper transport on an IP network

10. Which of the following does not reside in the call-processing agent UC model category?

A. Call agents

B. PBX agents

C. Voice gateways

D. Call control agents

11. Which signaling protocol that is compatible with Cisco IP phones is an IETF standard?

A. MGCP

B. H.323

C. SIP

D. SCCP

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12. Which of the following is not a signaling protocol that can be configured on voice gateways?

A. SPCP

B. MGCP

C. H.323

D. SIP

13. Which of the following Cisco routers cannot act as a Cisco voice gateway?

A. 2900 series ISR

B. 1800 series ISR

C. 1900 series ISR G2

D. 2600XM series

14. A 2900 Series ISR G2 with a voice gateway IOS and the proper modules can do all the following functions except what?

A. Conference call offloading

B. SRST

C. Connect analog phones

D. Emergency Responder offloading

15. At which segment of the Cisco Unified Communications model is QoS handled?

A. Network infrastructure

B. Applications

C. Call processing agents

D. Endpoints

16. Which of the following is not a consideration when choosing a voice gateway signaling protocol?

A. Call agent hardware and software

B. Legacy hardware used

C. Voice gateway hardware and software

D. Quality of Service requirements

17. What UC deployment model uses dispersed call agents that act as a single clustered voice system?

A. Centralized services model

B. Distributed services model

C. Inter-networking of services model

D. Geographical diversity model

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Review Questions 29

18. When using the distributed services UC deployment model, what additional feature is often recommended?

A. WAN links 5 Mbps or higher

B. SRST

C. H.323 signaling

D. MGCP signaling

19. Which UC deployment models recommend QoS on WAN links? (Choose all that apply.)

A. Inter-networking of services model

B. Centralized services model

C. Distributed services model

D. Geographical diversity model

20. When would a network administrator choose the inter-networking of services model over the other three UC deployment models?

A. If there is a single building or campus site and only a few small remote offices.

B. If there are several large dispersed campus sites and WAN links are slow and/or unreli-able.

C. If there are several large dispersed campus sites and WAN links are large enough to handle voice traffic.

D. If there is a single building or campus site and WAN links are large enough to handle voice traffic.

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Answers to Review Questions

1. D. The central offi ce is a geographically located offi ce that houses PSTN switch equipment.

2. A. The demarc is the point where the wiring responsibilities are split between the private owner and the phone company.

3. B. A tie trunk is the name used to describe a circuit that connects two PBX switches.

4. B. An interoffi ce trunk is the name used to describe a circuit that connects two PSTN switches located in separate COs.

5. A. Central offi ces today have one or more area codes assigned to them and they maintain pools of subscriber code numbers.

6. D. In the three-tiered PSTN hierarchy, CO-to-CO calling over interoffi ce trunks would be considered local calling. Long-distance charges would apply if a call needed to be sent to the interexchange network.

7. B. The Cisco SPA 300 and SPA 500 series phones do not support SCCP.

8. B. The Cisco Emergency Responder falls within the applications category of the Unifi ed Communications model.

9. B. Analog-to-IP adapters sit on the edge of an IP network and translate analog signaling into IP for proper transport on an IP network. This lets people continue to use analog telephone hardware on an IP network.

10. B. PBX agents are not one of the three Cisco call-processing agents as defi ned by Cisco.

11. C. Most current Cisco phones can run either SCCP or SIP signaling protocols. SCCP is Cisco proprietary while SIP is an IETF open standard.

12. A. SPCP is a modifi cation of the Cisco proprietary SCCP signaling protocol that is used only for communications between a CUCM Express call agent and Cisco SPA 300 and 500 series phones. Voice gateways cannot be confi gured with SPCP signaling.

13. C. The 1900 series ISR G2 cannot run voice IOS software and therefore can’t be used as a voice gateway.

14. D. The Cisco ISR can support numerous voice gateway functions, but it cannot offl oad Cisco Emergency Responder duties.

15. A. QoS is confi gured and maintained on network infrastructure hardware.

16. D. Quality of Service requirements are not a factor when choosing between voice gateway signaling protocols.

17. D. The geographical diversity model clusters call agents that communicate as a single unit over WAN links.

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Answers to Review Questions 31

18. B. SRST is recommended when deploying the distributed services UC deployment model. SRST is used when WAN links fail and remote sites need to make outbound calls.

19. C, D. The WAN is a critical component in the distributed services and geographical diversity models. Because these two models have voice traffi c sent across the WAN, QoS is therefore recommended.

20. B. The inter-networking of services model is used in situations where there are several dispersed buildings or campus sites and WAN links are not capable of handling voice traffi c.

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Answers to Written Lab 1.11. Central offi ce

2. Plant

3. Tie trunk, CO trunk, and interoffi ce trunk

4. Endpoints, applications, call-processing agents, and network infrastructure

5. Key system

6. Endpoints

7. Voice gateway

8. Call processing

9. Analog-to-IP adapter

10. Distributed services

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Documents

Chapter An Introduction to 1 Traditional Telephony …...An Introduction to Traditional Telephony and Cisco Unified Communications THE FOLLOWING CVOICE EXAM OBJECTIVES ARE COVERED