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CCVP CVOICE Quick Reference Sheets

Kevin Wallace

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[ 2 ]

ABOUT THE AUTHOR

CCVP CVOICE Quick Reference Sheets

by Kevin Wallace

About the Author

Kevin Wallace , CCIE No. 7945, CCSI, CCNP, CCDP, MCSE 4, CNE

4/5, is a full-time instructor for KnowledgeNet, a pioneer of next-

generation e-learning. With 15 years of Cisco internetworking experi-

ence, Kevin has been a network design specialist for The Walt Disney

World Resort and a network manager for Eastern Kentucky University.

Kevin holds a bachelor of science degree in electrical engineering form

the University of Kentucky. Among Kevin’s other publication credits

are the CCDA/CCDP Flash Cards and Exam Practice Pack and the

CCIE Routing and Switching Flash Cards and Exam Practice Pack ,

both coauthored with Anthony Sequeria and available from Cisco Press.

In addition, Kevin authored the Cisco Enterprise Voice over Data

Design (EVoDD) 3.3 course and has written for the Cisco Packet

magazine. Kevin also holds the IP Telephony Design Specialist and

IP Telephony Support Specialist CQS certifications.

About the Technical Reviewers

y Saa O #750

Jeremy D. Cioara , CCIE, MCSE 4/2000, CNE 4/5, is a trainer at

KnowledgeNet, a Cisco Learning Partner providing career Cisco

certification training. In addition to training in all major certification

programs (Cisco, Microsoft, and Novell) for more than seven years, he

has authored many books, including Cisco IP Telephony (CIPT) 3.3

Curriculum and Cisco IP Telephony Troubleshooting (IPTT) 3.3

Curriculum , and he attained the three Cisco IP Telephony Specialist

certifications. Outside of training, Jeremy has consulted for companies

such as Qwest, MicroAge, Terminal Processing Systems, and IKON.

Paul Giralt , CCIE R&S, CCIE Voice No. 4793, is an escalation engi-

neer at the Cisco Systems Technical Assistance Center in Research

Triangle Park, NC. He has been troubleshooting complex IP telephony

networks since the release of CallManager 3.0. Paul has troubleshot

problems in some of the largest Cisco IP Telephony deployments and has

provided training for TAC teams around the globe. He hold a bachelor

of science degree in computer engineering from the University of

Miami and is the author of Troubleshooting Cisco IP Telephony .

Jose Martinez , CCIE No. 1690, is an escalation engineer at the Cisco

Systems Technical Assistance Center in Research Triangle Park, NC.

Since 1995, he has worked in the TAC, supporting multiple technologies,

including IBM protocols, L2 and L3 switches, and most recently, IP

telephony. As an escalation engineer, Jose has been involved in numerous

network deployments. He has troubleshot problems involving critical

situations in customer’s networks and has provided training to other

Cisco engineers, partners, and customers.

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OVERVIEW OF LEGACY AND

IP TELEPHONY NETWORKS

CCVP CVOICE Quick Reference Sheets

by Kevin Wallace

Overview of Legacy and IP

Telephony Networks

switches [for example, Private Branch Exchanges (PBXs) or key

systems] located locally.

n Trunks interconnect phone switches.

Introduction

Understanding emerging Voice over IP (VoIP) technologies requires a

solid understanding of legacy telephony networks. Therefore, the Quick

Reference Sheets in this section review many of the terms and concepts

that surround legacy and packet telephony networks, and those tech-

nologies are then contrasted with IP Telephony technologies.

Companies that have their own phone switches can select between

PBXs or key systems. PBXs are typically more scalable than key

systems, supporting 20 to 20,000 phones. PBX users in the United

States typically dial a 9 to access an outside line. However, key systems

traditionally have buttons on a key phone that the user presses to access

a specific outside line. For example, you might have been in a store and

heard an intercom announcement such as, “Kevin, pick up line 2.” In

that example, Kevin would go to a “key phone” and press the line 2

button to access the call. Because of their scalability limitations, key

systems typically support a maximum of 30 to 40 users.

Call signaling makes it possible to place an end-to-end voice call.

Consider the following steps that are used to establish an end-to-end

voice call:

1. A phone goes off-hook and sends digits to the local phone switch.

2. The local phone switch examines the dialed digits, makes a

forwarding decision, and sends signaling information to the

destination phone switch.

3. The destination phone switch signals the destination phone by

sending ringing voltage to the phone.

Legacy Telephony Networks

The Public Switched Telephone Network (PSTN) is at the heart of the

legacy telephony network. PSTN components include the following

items:

n Edge devices (for example, phones) are used by customers to

interface with the PSTN.

n Local loops connect customer locations to a local central office

(CO) over a pair of wires called tip and ring .

n Phone switches allow one phone to connect to another phone by

dialing a phone number. The switch interprets the dialed digits and

interconnects the dialing phone’s local loop with the destination

phone’s local loop. The “phone company” has switches that are

located in COs. However, companies can have their own phone

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OVERVIEW OF LEGACY AND

IP TELEPHONY NETWORKS

CCVP CVOICE Quick Reference Sheets

by Kevin Wallace

When a user dials digits on an analog phone, those digits can be

communicated to the local phone switch using either dual-tone

multifrequency (DTMF) or pulse dialing. DTMF sends tones that

are composed of two frequencies, whereas pulse dialing rapidly

opens and closes the local loop to indicate dialed digits.

With digital circuits, such as T1s or E1s, multiple conversations can be

carried in different channels on the same circuit. Each of these digital

channels needs bits for signaling information. Common approaches

include the following:

n Common channel signaling (CCS) —Has a channel that is dedi-

cated to signaling. For example, in an Integrated Services Digital

Network (ISDN) circuit, the D channel is dedicated to signaling.

n Channel associated signaling (CAS) —Can use framing bits from

a few of the channels to serve as signaling bits. Sometimes this is

called robbed-bit signaling .

n E&M wink start —Seizes a line when the polarity on an E&M

circuit is reversed and then quickly flipped back to the original

polarity

Digital circuits can use multiplexing techniques to place multiple

conversations on a single link. For example, time-division multiplexing

can give a “time slice” to a specific channel, and by “taking turns,” you

can send 24 conversations across a single link.

Frequency-division multiplexing (FDM) allows multiple conversations

to be sent at the same time using different frequencies. For example,

dense-wavelength-division multiplexing (DWDM) simultaneously

sends multiple light frequencies over a fiber-optic cable.

Packet Telephony Networks

Many companies that have PBXs at more than one site and interconnect

those PBXs through the PSTN are migrating to a packet telephony

network. A packet telephony network allows companies to preserve

their existing investment in PBX technologies, while eliminating the

recurring expense for the trunks that interconnect their PBXs. Specifically,

companies can connect their PBXs to routers that are already interconnected

through a wide-area network (WAN). The PBXs can then send their

signaling information and voice calls over the WAN.

Call-forwarding intelligence can reside in the routers. For increased

scalability, however, you can configure routers to point to external call

Analog circuits have their own signaling mechanisms, such as the

following:

n Loop-start —Causes a phone switch to seize a line when loop

current is flowing

n Ground-start —Causes a phone switch to seize a line after the

phone temporarily grounds the “ring” side of the circuit

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OVERVIEW OF LEGACY AND

IP TELEPHONY NETWORKS

CCVP CVOICE Quick Reference Sheets

by Kevin Wallace

agents. Such a topology lays the foundation for other packet telephony

technologies, including the following:

n IP phones have an Ethernet connection that sends and receives

voice calls.

n Call agents replace much of the functionality that was provided

previously by the PBX. For example, a call agent can be config-

ured with route plans that dictate how voice calls are forwarded.

The Cisco CallManager (CCM) is an example of a call agent.

n Gateways can forward calls between different types of networks.

For example, a call from an IP phone could be forwarded through

a gateway to the PSTN.

n Gatekeepers keep track of WAN resources and, based on available

resources, either permit or deny a request to place a call across the

WAN. In addition, the gatekeeper can provide E.164 number reso-

lution.

n Multipoint Control Units (MCUs) contain digital signal processor

(DSP) resources and can support the mixing of audio streams in a

conference call.

IP Telephony Networks

Although packet telephony is more of a generic term, covering Voice

over IP (VoIP), Voice over Frame Relay (VoFR), and Voice over ATM

(VoATM), the primary focus of these Quick Reference Sheets is creat-

ing IP-based telephony networks using VoIP technologies. Therefore,

with the foundational understanding of legacy and packet telephony

networks, you delve into some of the components of IP Telephony.

First, you consider the analog interfaces that are available on voice-

enabled routers.

A Foreign Exchange Station (FXS) port allows you to connect plain old

telephone service (POTS) devices to a router. For example, you could

attach a traditional analog phone, speakerphone, or fax machine to an

FXS port on a Cisco router, and that FXS port can act like a PBX or

CO switch. For example, an FXS port can provide a dial tone when the

phone goes off-hook, interpret dialed digits, and send ringing voltage to

the attached phone.

A Foreign Exchange Office (FXO) port connects to a phone switch (for

example, a PBX or the PSTN). The FXO port can connect into the

traditional tip-and-ring connection that comes from a CO or a PBX.

Because it is acting as a phone, an FXO port can go off-hook, dial

digits, and answer incoming calls.

E&M is the third type of analog port, and this port interconnects PBXs.

The “E” and “M” originally referred to “earth” and “magneto,”

although you can think of “ear” and “mouth” to better visualize the

receive and transmit functions of E&M.

Simply placing a voice call across a WAN does not guarantee the

quality of the voice call. Data applications, for example, tend to be

more forgiving of dropped or delayed packets than applications such as

voice or video. Therefore, the quality of service (QoS) technology is an

integral part of Cisco VoIP designs, and an entire section in these Quick

Reference Sheets focuses on QoS technologies.

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