98162302 CCNA Voice Quick Refence Sheet

CCNA Voice Quick Reference

Michael Valentine

c i s c o p r e s s . c o m Your Short Cut to K n o w l e d g e

http://ciscopress.com

As a final exam preparation tool, the CCNA Voice Quick Reference provides a concise review of all objectives on the new IIUC

exam (640-460). This digital Short Cut provides you with detailed, graphical-based information, highlighting only the key

topics in cram-style format.

With this document as your guide, you will review topics on concepts and commands that apply to Cisco Unified

Communications for small and medium-sized businesses. This fact-filled Quick Reference allows you to get all-important

information at a glance, helping you focus your study on areas of weakness and enhancing your memory retention of essential

exam concepts.

About the Author Mike Valentine has 13 years of experience in the IT field, specializing in network design and installation. He is currently a

Cisco trainer with Skyline Advanced Technology Services and specializes in Cisco Unified Communications, CCNA, and

CCNP classes. His accessible, humorous, and effective teaching style has demystified Cisco for hundreds of students since he

began teaching in 2002. Mike holds a bachelor of arts degree from the University of British Columbia and currently holds the

MCSE: Security, CCNA, CCDA, CCNP, CCVP, IPTX, QoS, CCSI #31461, CIEH, and CTP certifications. He has completed

the CCIE written exam.

Mike was on the development team for the Cisco Unified Communications Architecture and Design official Cisco courseware

and is currently developing custom Unified Communications courseware for Skyline. Mike coauthored the popular CCNA

Exam Cram, second edition, first published in December 2005, as well as the third edition of that volume published in

December 2007.

About the Technical Editor Denise Donohue, CCIE No. 9566, is manager of Solutions Engineering for ePlus Technology in Maryland. She is responsible

for designing and implementing data and VoIP networks and supporting companies based in the National Capital region. Prior

to this role, she was a systems engineer for the data consulting arm of SBC/AT&T. Denise was a Cisco instructor and course

director for Global Knowledge and did network consulting for many years.

© 2008 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.

C C N A Voice Quick Reference by Michael Valentine

Introduct ion

I n t roduc t ion Voice over IP (VoIP) is no longer an interesting sidebar technology; it is a fact of day-to-day life for millions of people, some of whom are not even aware they are using it. Cisco has aggressively pursued the development and deployment of its Unified Communications suite of products and can now offer an integrated voice, video, and data solution for any business, whether it has just a few employees or a hundred thousand worldwide. The technology is reliable, user friendly, and exciting, but it is not simple—and a successful deployment requires that the designers, implementers, and administrators of a Unified Communications system know what they are doing. Training and certification of key staff are strategic components of any business plan to deploy a Unified Communications system. Until recently, the training track for Unified Communications went from the C C N A (the Associate-level routing and switching certification) straight to C C V P , the Professional-level voice certification. The transition between the certifications was difficult for many, because the C C N A did not examine any Unified Communications topics, and the C C V P launched directly into advanced VoIP signaling protocols, Unified Communications Manager administration, traditional telephony, gateway and gatekeeper configuration, QoS , and so on—all the while assuming that the student had a firm grasp of routing and switching concepts. I have met many good C C N A students who had no telephony or VoIP background and consequently had great difficulty in the C C V P program. Likewise, many students with very strong traditional telephony experience were quickly lost in the intensive data concepts of the C C V P curriculum. It was clear to me and to many of my colleagues that the C C N A was not a good fit as a prerequisite to C C V P . All this brings us to some good decisions that were made regarding Cisco Unified Communications training and certification. The C C N A has itself been split into C C E N T and C C N A , with C C N A serving as the foundation to some new and specialized certifications at the Associate level. The I I U C curriculum prepares students for the C C N A Voice certification, which in turn is a solid preparation for and a much-needed transition to C C V P .


CCNA Voice Quick Reference by Michael Valentine

In t roduc t ion

Purpose of This Guide This document serves as a roadmap of the CCNA Voice curriculum and a quick reference for the concepts and commands

that apply to Cisco Unified Communications for small and medium-size businesses. This document is not a list of all the

questions you may be asked on the exam, but you can be sure that the exam will touch on all the topics you find here.

Reviewing this document should help you remember key points and commands you will need to know for the exam.

Who Should Read This Guide Anyone who is preparing to take the CCNA Voice exam will find this guide useful. Some may use it as in introduction,

and some as a refresher right before their test, some perhaps both. Data networkers who need a quick but complete intro

duction to Cisco Unified Communications for a small or medium-size business will find it useful as well. Those of you

who are getting back into study mode for a CCVP exam may turn to this guide as a refresher, too. Then there are always

those who simply want to learn something new.

Whoever you are, welcome and enjoy the text. I hope you find it useful.

Introduct ion to Unif ied Communicat ions Today's work environment can be very different from what our parents experienced. The business environment is more

competitive, with an unrelenting pressure to be more efficient, to react quickly, and to make important decisions instantly.

Efficiencies can be gained by reducing costs, which in turn increases profit, but significant gains can also be made by

investing in the business infrastructure so that productivity increases dramatically. Increased productivity means more

opportunities to profit from a newfound competitive edge. This is known as Return on Investment, or ROI. The goal is to

maximize the ROI—for every dollar spent, businesses want to see more dollars earned, or at least fewer dollars wasted.



In t roduc t ion

One area in which businesses have found ways to improve their ROI is in their communications. The evolution of

communications from traditional telephony, through cell phones, to smart phones and email, and now to Unified

Communications, has created opportunities for businesses to access information and get it to workers instantly. Unified

Communications puts voice, data, and video on a converged single network. This makes monitoring, administering, and

maintaining the network simpler and more cost effective than if three separate systems existed. Unified Communications

also puts powerful applications with information-distribution features right where they are needed. Workers today can be

almost anywhere and can carry out meaningful or even critical tasks anywhere they can get a connection to the converged

network.

The next significant feature of a Unified Communications system is that it is easy to scale, adding more users, more loca

tions, and even more features. Because the Cisco Unified Communications system is a distributed collection of devices,

functions, and features that are linked by common protocols, adding a new component is much simpler, and integration of

the new component's capabilities and features can appear seamless to the people who use the system.

The components required to create and use such a system are numerous and complex. Cisco has taken significant steps to

develop, document, release, and support the various components as an integrated system. The next section examines the

components of a Unified Communications system and introduces the devices and applications that make up the system.


F IGURE 1

The Unified

Communications

Architecture



• Infrastructure Layer: This layer refers to the network itself, made up of connected switches, routers, and voice

gateways. This is the converged network that carries data, voice, and video between users on the system.

• Call-Processing Layer: This layer manages the signaling of voice and video calls. When a user picks up the phone

and dials a number, the call processing agent determines how to route the call, instructs the phones to play dial tone

or to ring, and records the details of the call for future analysis. The call agent carries out many other functions; it

can be considered the equivalent of a traditional PBX system, but with many more features.

• Applications Layer: This layer features elements such as voice mail, call-center applications, billing systems, time-

card or training systems, and customer resource management applications—to name just some of the many applica

tions that can integrate with, draw from, or otherwise complement the Unified Communications systems. Because

the Unified Communications systems are distributed (meaning not constrained to one box or even one location), the

applications can be hosted almost anywhere, given appropriate connectivity.

• Endpoint Layer: This layer includes the parts of the system that the users see, hear, or touch. This includes Cisco

Unified IP Phones, PCs with software phones, video terminals, or other applications that send and receive informa

tion from the Unified Communications system.

The following sections examine the layers in a little more detail.

Infrastructure Layer At the infrastructure layer, we are building the connections between all the devices that send and receive data, voice, and

video. These include Layer 2 and 3 switches, routers, and voice gateways. Voice gateways are among the most important

components because they provide the connection to the PSTN or other network carriers. One of the critical functions (and

one that is unfortunately often underemphasized in many deployments) is quality of service, or QoS. QoS provides

service guarantees to various types of network traffic, in particular voice and video traffic. Without QoS, you can experi

ence poor call quality or even failed calls. Infrastructure design and deployment is literally the foundation of the system;


if any weaknesses exist here, they will manifest as system failures or unreliability. It is very important to build a solid and

correct foundation. The goal is to achieve 99.999% uptime; achieving that goal takes careful attention and good design.

Call Processing Layer The call processing layer is chiefly about the call agents. A call agent is not a person; it is an application that looks at the

signaling traffic from devices that place and receive calls, and it determines what to do with the call. A Unified IP Phone

sends a packet to the call agent when you lift the receiver; the call agent instructs the phone to play a dial tone. When you

begin dialing a number to call, the call agent receives the digits and tries to find a match for the number in its dial plan. If

the destination number is a phone that it controls, it tells the called device to ring. During the call, the call agent also sets

up other services, such as Hold, Call Park, Transfer, Conference, and so on. The call agent also instructs the phones to

tear down the call when one party hangs up. The call agent usually keeps detailed records of each call made; these are

commonly used for billing purposes or troubleshooting.

Cisco provides several options for call agents, matched to the size and requirements of the customer:

• The Cisco Smart Business Communications System is designed for small businesses with up to 48 users. The system

runs on the Cisco Unified Communications 500 Series for Small Business devices.

• Cisco Unified Communications Manager Express serves up to 240 users and runs on the Integrated Services Router

platforms.

• Cisco Unified Communications Manager Business Edition handles up to 500 users and runs as a standalone installa

tion on a 7800-series Media Convergence server.

• Cisco Unified Communications Manager can handle 30,000 or more users and runs on clusters of 7800-series Media

Convergence servers.

The Smart Business Communications System is a group of specially designed, integrated devices that can provide high-

quality routing, firewall, intrusion prevention, Power over Ethernet, wireless, and many WAN and PSTN connectivity

options. It is essentially a solution-in-a-box, with a simple web-based interface that is largely plug and play. The Unified

Communications 500 Series devices are small and inexpensive, providing the kind of connectivity options small busi

nesses need to allow them to take advantage of Unified Communications with a good ROI. The SBCS is expandable

using 500-series switches, and the call agent software can support up to 48 phones. Voice mail and Auto-Attendant func

tions are provided by the integrated Cisco Unity Express application.


FIGURE 2 The Smart Business Communications System—Image © Cisco

Smart Business Communications System

Unified Communication Manager Express

F IGURE 3 Cisco Integrated Services Routers for Unified

Communications Manager Express— Image © Cisco

Cisco Unified Communication Manager Express is a software feature that can run on the ISR-series router platforms,

including the 800, 1800, 2800, 3800, and 7200-series platforms. The call agent application is embedded with the Cisco

IOS software and is configured either from the command line or a Web-based interface. Unified CM Express is a full-

featured call agent that is cost-effective, reliable, and scalable and integrates with both Service Provider connections and

Unified Communications Manager clusters. With support for both H.323 and SIP protocols, site-to-site connections are

possible in a variety of environments. The Unified CM Express system can be set up either as a PBX or a Key switch

system, providing customers with a familiar experience that suits their operating environment.

Unified Communications Manager, Business Edition Unified Communications Manager, Business Edition is a standalone installation of the Unified CM application and Cisco

Unity Connection, coresident in a single MCS 7800-series appliance. This system can support up to 500 users in a single

site or multisite centralized deployment and can be migrated to a full CM cluster if growth necessitates it. Unified CM

Business Edition provides medium-size businesses with advanced features such as Mobility (a.k.a. Single Number



Reach), Do Not Disturb, Intercom Whisper, and Audible Message Waiting Indication, as well as speech recognition and

integrated messaging. Because Unified CM Business Edition uses the same call agent software as a full cluster deploy

ment of Unified CM, it supports full integration with the other Unified Communications applications, such as Unified

Presence, Unified Personal Communicator, MeetingPlace Express, Contact Center Express, and so on.

Unified Communications Manager The full version of Unified Communications Manager is an enterprise-class, fully scalable, redundant, and robust distrib

uted packet-telephony application. Scalable to 30,000 users per cluster, with the capability to form intercluster connec

tions, it can support a global unified communications system for hundreds of thousands of endpoints. Unified CM

versions prior to 5.x are Windows based, whereas versions 5.x and 6.x are Linux-based appliances.

Applications Layer There are effectively a limitless number of applications that can be part of a Unified Communications system, because

third-party applications can be developed to closely integrate with the Cisco suite of products. The following is a list of

the more common applications found in a Unified Communications system:

• Voice Mail: Voice mail can be provided using Cisco Unity, Unity Connection, or Unity Express. Unity and Unity

Connection run on the MCS 7800 series platforms, and Unity Express is a self-contained module that is added to an

ISR router and administered through the command line and GUI. The maximum mailboxes and recording time

capacities vary depending on which module (either Advanced Integration Module or Network Module) is installed in

the router.

• Cisco Emergency Responder: This application tracks the location of an IP telephony device based on the physical

switch port it is connected to. This information is attached to the caller information in the event the device calls 911,

which in turn allows 911 responders to locate the device (and therefore presumably the emergency) more precisely.

911 operation in a Unified Communications environment is a major design challenge because a VoIP phone system

can easily throw out the premise that a PSTN call is placed from the same location as the phone that made it.


• Cisco Unified Contact Center [Express]: This is a call center application with full feature support for advanced

call distribution, supervision, escalation and logging. Versions are available to support small and large call centers.

• Cisco Unified Meeting Place [Express]: This is a full-featured web-conferencing application enabling voice and

video conferencing as well as document sharing and collaboration, whiteboarding, and conference participant

management.

• Cisco Unified Presence: This extends the native capabilities of Unified CM 6.x+ to indicate presence information.

The native capability includes on/off hook status in speed dials and call lists, whereas the full applications server

provides detailed presence information as typically found in chat applications ("On the Phone," "Out to Lunch," "Do

Not Disturb," and so on).

Endpoints Layer An increasing variety of Cisco Unified IP Phones (and third-party IP phones) can be part of a Unified Communications

deployment. All Cisco Unified IP Phones provide a display-based user interface, user customization, Power over Ethernet

capability (where appropriate), and support for G.711 and G.729 codecs (and, on some models, Cisco Wideband and/or

iLBC codecs). The following is a partial list and brief description of the Cisco Unified IP Phones available:

Commercial/Retail Phones

7931G: 24 programmable buttons, 4-way LEDs, Dedicated HolaVTransfer/Redial buttons

7921G: Wireless, 2-in. color screen, speakerphone, XML-PTT, longer battery life

Mobility

7921G: Wireless, 2-in. color screen, speakerphone, XML-PTT, longer battery life

IP Communicator: Software-based IP Phone, emulates 7970G functionality


Business Class

7940G: B/W LCD, 2-button, XML-capable, SIP-capable

7941G: Higher resolution B/W LCD, 2-button, XML-capable, SIP-capable

7960G: B/W LCD, 6-button, XML-capable, SIP-capable

7961G: Higher resolution B/W LCD, 6-button, XML-capable, SIP-capable

Advanced Media

7942G: Hi-fidelity audio, Hi-res display, 2-button, XML-capable, SIP-capable

7945G: Gig Ethernet, Hi-fidelity audio, Hi-res color display, 2-button, XML-capable, SIP-capable

7962G: Hi-fidelity audio, Hi-res display, 6-button, XML-capable, SIP-capable

7965G: Gig Ethernet, Hi-fidelity audio, Hi-res color display, 6-button, XML-capable, SIP-capable

7975G: Gig Ethernet, Hi-fidelity audio, backlit hi-res color display, 6-button, XML, SIP-capable

Color Touch

7970G: Backlit hi-res color touch screen, 8-button, XML-capable, SIP-capable

7971G-GE: Gig Ethernet, Backlit hi-res color touch screen, 8-button, XML-capable, SIP-capable

7975G: Hi-fidelity audio, Backlit hi-res color touch screen, 8-button, XML-capable, SIP-capable

Video

7985G: Personal desktop video phone

Unified Video Advantage: Software IP Video Phone with support for attached camera

Conference

7936G: Backlit LCD, 3 softkeys, small-medium conference needs

7937G: Hi-fidelity audio, extended audio coverage w/ extra mics, large display

Messaging A variety of messaging options are available to suit the needs of businesses small and large. The following table provides

a summary of the options.

Product Max. Users Messaging Capability Platform

TDM PBX Integration? Networking? Redundancy?

Unity Express 250 Voice Mail + Integrated Messaging ISR No Yes No

Unity Connection 3000 Voice Mail + Integrated Messaging MCS Yes No No

Unity 7500 per server

Voice Mail + Integrated Messaging + Unified Messaging

MCS Yes Yes Yes

The following sections describe the messaging products listed in the table in more detail.

Cisco Unity Express Unity Express is an ISR-based application that runs either on an AIM module or an NM module. AIM modules are

connected to the main board as a daughter board addition and use flash memory for greetings and message storage. AIM

modules therefore have less capacity for storage. NM modules are inserted into module bays in ISR routers, use a hard


Understanding Unif ied Communicat ions Appl icat ions In this section, we examine the variety of applications available for integration in a Unified Communications environ

ment, including Messaging, Auto Attendant, Interactive Voice Response (IVR), Contact Center, Mobility, and Presence.


disk for greeting and message storage, and have greater capacity for storage than AIM modules. Unity Express supports

from 4 to 16 concurrent sessions and 12 to 250 mailboxes (dependent on the module and platform installed). Unity

Express is managed through the command line or a web-based GUI. It allows users to view and sort their voice messages

using the IP Phone display, email application, or IMAP client. Unity Express can be deployed in conjunction with

Unified CM or CM Express and can supplement a full Unity deployment.

Cisco Unity Connection Unity Connection is a medium-size business solution with a full range of messaging features. It can be deployed on its

own or as a coresident installation as part of Unified Communications Manager Business Edition on suitable MCS plat

forms. When deployed as part of CM Business Edition, Unity Connection supports up to 500 users; when deployed as a

standalone application, Unity connection supports up to 3000 users per server (dependent on hardware). Scalability is

achieved by networking up to 10 other Unity messaging products of any type. Fourteen languages are supported for

deployments worldwide. Unity Connection also supports speech recognition, allowing users to speak commands to

manage their messages hands-free. Multiple interfaces are supported for managing messages from an IP Phone, an email

client, a web GUI, or Cisco Unified Personal Communicator. Users can define their own rules to transfer calls based on

caller, time of day, and Microsoft Exchange calendar status.

Cisco Unity Unity is the enterprise-class messaging application with support for up to 7500 users per server and up to 250,000 users

in a multi server networked environment. Interoperability with legacy voice-mail systems, notably Octel and Nortel

systems, allows a phased transition to IP messaging with minimal disruption to users. Unity supports 35 languages, facili

tating deployments worldwide. Full unified messaging is possible with connectors for Exchange, Notes, and GroupWise,

providing a single inbox for email, voice mail, and fax messages. Text-to-speech capability allows users to have their

emails read to them over the phone by the RealSpeech engine; speech recognition is also available so users can instruct

Unity to play, search, or record messages hands-free. Secure messaging is supported, allowing encrypted messages and

preventing messages that have expired from being played. Access to messages is made simple, intuitive, and possible

from almost anywhere.


Auto Attendant An Auto Attendant is basically an advanced answering machine; instead of only one message, it can play several, depend

ing on the date and time, which number was called, and most importantly, what numbers the callers pressed in response

to the greeting they heard. If you have ever heard: "For service in English, press I. Pour service en Francais, appuyez sur

le 2 . . . , " you have been served by an Auto Attendant. Typically, Auto Attendants allow callers to select the department or

extension they want to call, and often they allow the caller to spell out a first or last name to search in the company direc

tory. Cisco Unity, Unity Connection, and Unity Express all provide Auto Attendant functionality; Unity and Unity

Connection include a simple web interface that makes it very easy to construct menus and test to see that they work as

you intended.

Cisco Unified IP IVR Although Auto Attendants are useful, their functionality is limited to pretty basic menu navigation. To scale this function

ality up to call-center size, and especially to include speech recognition, prompt-and-collect ("Please enter your 10-digit

account number, followed by the # sign"), Text-to-Speech, database integration, and Java application integration, a much

more advanced IVR application is required. Cisco Unified IP IVR has all these advanced capabilities. Call centers that

have a high call volume and many possible queues of callers waiting for different agent capabilities can effectively deploy

Unified IP IVR to steer callers to the correct agent, or perhaps to an automated information source without the need to tie

up an agent at all. Unified IP IVR includes the capability to provide both real-time and historical reports on its utilization

and offers multiple-language support.

Cisco Unified Customer Voice Portal For the very largest call centers, the Unified CVP product provides advanced IVR, including speech recognition,

advanced queuing, integration with Cisco Unified Contact Center (Enterprise and Hosted), and powerful call routing,

management, and reporting features.


Cisco Unified Contact Center Cisco provides a range of Contact Center products for SMB, Enterprise, and Service Provider applications. Customer

contact solutions provide multiple avenues to reach and interact with customers, including basic telephony as well as

feature-rich web, email, and even video interaction. The three Contact Center products are described next:

• Cisco Unified Contact Center Express: Suitable for 10 to 300 agents, it provides sophisticated call routing,

outbound dialing capabilities, comprehensive contact management, and chat and web collaboration in a single-

server, integrated "contact center in a box."

• Cisco Unified Contact Center Enterprise: Provides intelligent contact routing, call treatment, network-to-desktop

computer telephony integration (CTI), and multichannel contact management. It combines multichannel automatic

call distributor (ACD) functionality. Sophisticated monitoring allows customers to be routed to the most appropriate

agent (based on real-time conditions such as agent skills, availability, and queue lengths) anywhere in the enterprise,

regardless of the agent's location.

• Cisco Unified Contact Center Hosted: An application hosted by service providers, who then lease its functionality

to customers who want a virtual contact center without the need to manage and maintain it themselves. Subscribing

business customers can have IP or time-division multiplexing (TDM) infrastructures or a combination of the two.

Contact Center Hosted provides all the advanced capabilities found in Contact Center Enterprise.

Cisco Unified Mobile Solutions Today's workforce is mobile, distributed, and utilizes multiple technologies to communicate. The desire to have a seam

less transition between the various ways in which people can be reached has spurred the development of mobility features

in Cisco Unified Communications. The key products are the following:

• Cisco Unified Mobility: (a.k.a Single Number Reach) Allows multiple remote destinations (commonly a cell phone,

a home office phone, or other work location) to be configured to ring at the same time as the worker's enterprise

desk phone. Thus, when a customer calls your work number while you are on your way to a meeting, your cell

phone can ring and you can answer without the customer realizing you are away from your desk. Furthermore, if

you return to your desk, you can simply pick up your desk phone and continue the call.

A related feature, called Cisco Mobile Voice Access, allows users to place calls from their enterprise desk phone

from a remote location or a cell phone. By dialing a configured number and entering an access code, the enterprise

system will prompt for the number you want to call, and the call will be placed as if you were at your desk. This is

useful not only for presenting the preferred Caller-ID number to the customer, but also potentially for long-distance

toll savings.

• Cisco Unified Personal Communicator: A desktop PC (or Mac) application that combines a software IP Phone, IM

client, video, and online collaboration capabilities. Presence indications ("Busy," "In a call," "Away," "Do Not

Disturb," and so on) can save time and enhance productivity because users can see the status of the person they want

to contact before trying to reach them. Integration with an Outlook toolbar provides click-to-call or click-to-chat

from a message or contact.

• Cisco Unified IP Communicator: A fully functioned software IP Phone, often characterized as a "7970 under

glass." Users can place and receive calls from their PCs from anywhere that connectivity to the call agent can be

established. This is typically achieved through a VPN connection; it is perfectly possible to place a call from an

airport boarding lounge or your local coffee shop. Unified IP Communicator can be enhanced with Unified Video

Advantage, which integrates a PC webcam for video calls.

• Cisco Unified Mobile Communicator: An application for smart mobile phones that provides access to enterprise

directories, presence indicators, secure text/chat, voice-mail access, call history of any of the user's phones displayed

on the mobile handset, and collaboration and conferencing integration with Unified Meeting Place.

• Cisco Unified Presence: A server-based application that extends the on/off hook status monitoring capability of Unified

CM 6.x to include IM-like status messages. Status indications can be displayed or integrated with Personal Communi

cator, Mobile Communicator, IP Phone Messenger, the Microsoft Office Connector, and IBM Sametime Communicator.


F IGURE 4 The Cisco Telepresence 3000 System—Image © Cisco

n i f ied C o m m u n i c a t i o n s A p p l i c a t i o n s

Cisco Telepresence Cisco Telepresence is a state-of-the-art high-definition videoconferencing system. A specially designed system of furniture,

cameras, monitors, and microphones creates a life-sized illusion of a meeting whose participants may be half a world apart.

With 1080p HD video, CD-quality spatial audio, and high-quality lighting, the experience is dramatic to say the least. In

combination with the Telepresence Multipoint Switch, up to 36 locations can be included in a single conference with near-

zero latency. This can only be described as a high-end solution, with commensurate demands on bandwidth.


the Public Switched Telephone Network (PSTN)

The PSTN, or Public Switched Telephone Network, is made up of Central Office switches to which subscriber lines are

connected. The CO switch is programmed so that it knows which phone number (subscriber line) is attached to a particu

lar port. If the number called is not on the local switch, the call is routed over an interoffice trunk to another switch,

which may have the called subscriber line connected directly to it or may in turn route the call to other CO switches.

Telephone numbering plans are organized so that calls are routed efficiently through the switch system to the correct

destination switch.


Understanding Tradi t ional Telephony This section introduces traditional telephony systems, concepts, and applications.

The PSTN

FIGURE 5 Public Switched Telephone Network

A Representation of

Note that for our purposes, a line connects to a single phone number and supports one call at a time, whereas a trunk

interconnects two switches and supports multiple calls at a time.

Business Telephony Systems Businesses have more elaborate requirements of the telephone beyond simply placing calls. Over time, two main types of

business systems have evolved: the PBX and the Key System. Both have their place, and both offer calling features that

make it easier to carry on business both internally and externally with staff, customers, and suppliers.

PBX Systems

F IGURE 6 A Representation of a PBX System

Business telephone systems often use a Private Branch Exchange (PBX) switch, usually located in their building. The

PBX is configured in much the same way as the PSTN CO switch: it holds the dial plan for all numbers within the busi

ness, and external calls are routed over a CO trunk to the PSTN CO switch if the called number is not on the PBX. As a

business grows, it is common to install another PBX in another location or building and set up a special trunk (called a

tie-line or tie-trunk) between the PBXs so that calls to the remote location are still internal numbers (typically 4- or 5-

digit numbers) instead of PSTN calls.

A PBX consists of a control plane (the "brain"), a terminal interface that connects phones to the features they want to use,

a switching engine that determines which port to route a call out, line cards to connect to phones, and trunk cards to



Unders tand ing Trad i t iona l Te lephony

connect to the PSTN or to tie trunks to other PBXs. PBXs come in a variety of sizes, supporting from 10 to 20,000

phones. All PBXs offer basic calling features, with additional advanced features optional based on hardware capability

and licensing. These features typically include Hold, Transfer, Conference, Park, Voice Mail, and so forth.

Key Systems Smaller businesses will sometimes use a key system. A key system is like a PBX in that it controls a group of local

phones, but key systems tend to have fewer features than PBXs. One characteristic of key systems that many businesses

specifically request is distributed answering from any phone; that is, all the phones ring at once, and whoever is able to

pick up Line 2 (for example) can push the Line 2 button on any phone and take the call. PBXs don't normally do this;

they have a central answering point (a receptionist or Auto Attendant) and Direct Inward Dial numbers (DIDs) if needed.

Telephony Signaling Telephony signaling refers to the messages that must be sent to set up and tear down a phone call—that is, anything other

than the actual voice. Following are the three types of telephony signaling:

• Supervisory: Communicates the current state of the telephony device. There are three types of supervisory signals:

• On-Hook: The phone is hung up. Only the ringer is active in this state. (Note that if the speakerphone button is

pressed, this is the same as being off-hook.)

• Off-Hook: The phone receiver is out of the cradle. This signals the phone switch (PSTN, PBX, or Key) that the

phone wants to make a call; the switch sends a dial tone to indicate that it is ready to receive digits.

• Ringing: The switch sends voltage to the phone to make it ring, alerting the user that there is an inbound call.

The other end of the call hears a ringback tone.

• Address: Communicates the digits that were dialed. Address signaling is most commonly done using Dual Tone

Multi Frequency (DTMF) tones, commonly known as TouchTone dialing. The combination of tones tells the switch

what number was pressed. Older systems also support pulse dialing, which is what the old-fashioned rotary dial




Unders tand ing Trad i t iona l Te lephony

phones used. Pulse dialing works by repeatedly opening and closing the circuit to the phone switch; the switch

counts the number of pulses and interprets that as the number dialed. You might have seen in really old movies

when someone picks up the phone and taps the receiver cradle repeatedly; this was how you got the attention of the

operator.

• Informational: Communicates the call status to participants in the call. Informational signals include dial tone, ring-

back tone, and reorder tone. These tones, and others not mentioned here, will vary from country to country. In

England, for example, ringback tone sounds very different from what would be heard in North America.

Signaling System 7 (SS7) SS7 is a global telephony standard that allows a phone call to be routed between CO switches, between long-distance

carriers, and even between national telephone providers in other countries. SS7's primary role is to complete the setup

and teardown of phone calls; this is quite a distinct process from the actual transport of the voice signal. In fact, the call

control information in an SS7 network must traverse an entirely separate network from the voice path. The capabilities of

SS7 have allowed the introduction of relatively complex value-added services, such as call screening, number portability,

and prepaid calling cards.

PSTN Call Setup To make a PSTN call, several steps occur that the caller is unaware of. The following steps refer to Figure 7.

FIGURE 7 PSTN Call Setup 0

Customer Telephone

FIGURE 7 PSTN Call Setup

NOTE Codes do not always need to be dialed; Local numbers must always be dialed.

Numbering Plans A numbering plan is an organized distribution of telephone numbers administered by a regional or national authority. The

plan defines the rules that allocate numbers according to an established international telecommunications standard. For

example, the North American Numbering Plan defines a country code of 1, followed by a three-digit area code, a three-

digit office code, and a four-digit local number. There are numerous other numbering plans for other countries or regions

of the world.


1. The calling phone goes off-hook, closing the circuit to the local CO switch.

2. The local CO switch detects that current is flowing over the closed circuit and sends a dial tone to the calling phone.

3. Address signals (DTMF or pulse) are sent as the calling party dials the called number.

4. The local CO switch collects the digits and makes its routing decision; in this example, it uses an SS7 lookup to

locate the destination CO switch.

5. Supervisory signaling indicates to the far-end trunk that a call is inbound.

6. The PBX determines which internal line the call should go to and causes the connected phone to ring.

7. The ringback tone is heard at the calling party end.

8. The called party goes off-hook, and a voice circuit is established end-to-end.

The fact that all this happens with very high reliability billions of times every day is pretty impressive. It also provides

some insight into how complex it is to duplicate these functions in a VoIP system. More on that later.

NOTE Several other ranges are reserved for specialized purposes. One commonly recognized one is 555-01XX, which is used in film and TV, demonstrations, or education. No actual customer is assigned these numbers, so calling a number seen in a movie will not pose a nuisance to anyone. When Tommy Tutone recorded "867-5309/Jenny," he immediately annoyed thousands of phone customers worldwide.

It is very important to note that the "N" represents any digit in the range 2 through 9, and the "X" represents any digit 0

through 9. You will never find an office or area code of OXX or 1XX; those numbers are either reserved for specialized

purposes or would interfere with things like operator access numbers. Several ranges are also reserved for Easily

Recognizable Codes (ERCs); these are numbers where the second and third numbers of the area code are the same. They

are used for special services—for example, 800, 888, 877, and 866 are toll-free numbers. Another recognizable assign

ment is the " N i l " series: this includes 411 , 611, and 911 numbers that are not used as area codes but for other special

assignments, such as information or emergency services.

E.164 Addressing The E.164 addressing scheme is an international standard for telephone numbering plans, originally developed by the

International Telecommunication Union. An E.164 number contains the following components:

CC—Country Code

NDC—National Designation Code

SN—Subscriber Number

An E.164 number is standardized at 15 digits, generating over 100 trillion unique strings. In theory, it's possible to direct

dial any conventional phone in the world from any other conventional phone.


The North American Numbering Plan Let's look at the NANP in more detail. The 10-digit number is made up of the 3-digit area code, the 3-digit office code,

and the 4-digit local number, as shown here:

NXX-NXX-XXXX


Introduct ion to Analog Circuits Analog (in contrast to digital) circuits are still the most common telephone connections worldwide. The phone line to a

North American home is most commonly an analog loop circuit, although more and more digital phone services are being

installed. Cisco gateways must connect to various analog services to place calls to the PSTN; the analog circuits that

Cisco supports are Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), and Earth and Magneto (E&M).

This section examines the components of an analog telephone and the signaling methods used by analog circuits.

Components of an Analog Phone An analog phone includes the following components:

• Receiver: The handset speaker

• Transmitter: The handset microphone

• 2-wire/4-wire hybrid: Converts 2-wire from the CO to 4-wire in the phone

• Dialer (tone/pulse): The dialing keypad or rotary dial

• Switch hook: The switch that closes/opens the circuit (off-hook/on-hook)

• Ringer: Sounds to indicate inbound call

Foreign Exchange Station An FXS port connects directly to an analog phone or fax machine. Switches (including CO switches and PBXs) and

Cisco gateways will have FXS ports to connect an analog phone. The switch or FXS gateway port must provide power,

call progress tones, and dial tone to the analog device. An FXS port on a gateway is also the direct connection to the VoIP

network and consequently also contains a coder-decoder (Codec) to convert the analog signal to digital for packetization.

Alternatively, a Cisco Analog Telephony Adapter can serve as a remote FXS-to-Ethernet converter to connect an analog

station to the VoIP network.

Foreign Exchange Office An FXO port connects to the PSTN CO switch. If you want to connect your gateway router to the phone company over

standard analog lines (that you could plug your analog phone into), you use FXO ports. These ports allow the gateway to

place and receive calls to/from the PSTN. FXO ports also include a codec.

F IGURE 8 Loop-Start Signaling

Loop-start signaling is commonly associated with local loop circuits (such as an analog line to the PSTN); it is seldom

seen on trunk connections. A local loop is a two-wire service that uses very simple electrical signaling; remember that

this technology has been in use and substantially unchanged for 100 years!



Following is the loop-start process:

1. A phone that is on-hook breaks the electrical circuit; we say opens the circuit. No electricity can flow because of the

open circuit.

2. When the receiver is lifted, the circuit closes and electricity flows. This current is - 4 8 V DC. The CO switch that is

connected to the local loop detects the current flow and interprets this as an attempt to place a call—we say "seize a

circuit." The CO switch plays dial tone down the line to the phone as an indication that it is prepared to collect

digits.

3. If the phone is on-hook and the CO switch has a call inbound for it, the CO switch applies 90V AC current to the

open circuit; because it is AC, the current can be applied even on the open circuit. By the way, this is why you

should not have an analog phone near the bath. The DC voltage won' t do much, but you will definitely know it if the

phone rings and you get zapped by the AC voltage.

Loop-start works very well for homes or other lightly used circuits, but if it is in constant use, a problem known as

glare can occur; this refers to both ends of the circuit being seized at the same time, so that you pick up the phone

and there is a caller on the line at the same moment, by coincidence.

Ground-Start Signaling Ground-start signaling is an adaptation of loop-start. Instead of the circuit being closed only at the phone end, both ends

of the circuit have the capability to detect current, and both ends can request and confirm the use of the circuit. This is

achieved by both ends being able to ground one of the wires in the circuit. These wires (or leads) are referred to as Tip

and Ring. These terms date back to the use of 1/4-inch jacks with a positive contact at the tip and a negative conductor in

the ring. The advantage is that it makes glare much less likely, and consequently ground-start is appropriate for trunk

connections that are heavily used. However, it is very rare to see a ground-start trunk in a VoIP network or indeed in any

new trunk deployment.

F IGURE 9 Ground-Start Signaling

The ground-start process as it occurs on a trunk between a PBX and the CO switch is described next; refer to the diagram

for each step:

1. The PBX has a call that it must send to the PSTN. It signals to the CO switch that there is an inbound call by

grounding the ring lead.

2. The CO senses the ring lead as grounded and grounds the tip lead to signal the PBX that it is ready to receive the

call.

3. The PBX senses the tip ground and closes the loop between tip and ring in response; the PBX also removes the ring

ground.

4. The voice circuit is complete, and communication can begin.



E&M Signaling Variously called "Ear and Mouth," "RecEive and TransMit," and "Earth and Magneto," E&M analog trunks were typi

cally used to interconnect PBXs (tie-trunks). E&M connections have separate leads for signaling and voice; the signaling

leads are known as the E and M leads.

In an E&M connection, one side is called the trunk side; this is usually the PBX side. The other side is called the

signaling-unit side; this is the CO, channel-bank, or Cisco gateway E&M interface. The E lead is used to indicate to the

trunk side that the signaling-unit side has gone off-hook; conversely, the M lead is used to indicate to the signaling-unit

side that the trunk side has gone off-hook.

Five types of E&M signaling exist, numbered Type I through Type V. In a Cisco Gateway application, Types II and V can

be connected back-to-back and Type I cannot be. Cisco does not support Type IV.

Three main techniques are employed in E & M circuit signaling:

• Wink Start: The terminating side (for example, a Cisco Gateway) uses a brief off-hook-on-hook "wink" to

acknowledge that the originating side (for example, a PBX) has gone off-hook. Upon receipt of the wink, the origi

nating side begins sending digits. When the far-end device answers the call, the terminating side goes off-hook and

the voice circuit is then set up.

• Immediate Start: The originating side goes off-hook, waits a set time (perhaps 200ms), and then begins sending

digits whether or not the terminating side is ready.

• Delay Dial: Assume that a PBX is placing a call outbound to the PSTN: First, the PBX goes off-hook. The CO then

goes off-hook until it is ready to receive digits; it then goes on-hook. (This time period is the delay dial signal.) The

PBX sends digits. When the far-end device answers the call, the CO goes off-hook (called Answer Supervision), and

the voice circuit is then set up. The advantage of Delay Dial is that some equipment is not ready to receive digits

instantly, even though it has sent the wink; the delay compensates for this.


Introduct ion to Digital Circuits Digital circuits have the chief advantage of allowing a much higher density of calls on a given physical connection; an

analog circuit can handle only one call at a time, whereas a digital circuit can handle many.

There are two main types of digital circuits: Common Channel Signaling (CCS) and Channel Associated Signaling

(CAS). CAS circuits are available in two speeds: Tl at 1.544Mbs supports 24 calls, and El at 2.048Mbs supports 30

calls. (For these values, we are assuming the calls are not compressed; more on this later). CCS circuits are designated as

PRI T l , PRI E l , and BRI. A PRI Tl can support 23 calls, a PRI El 30, and a BRI only 2.

The use of a digital circuit by definition implies that the voice signal must be digitized; the conversion from analog to

digital is performed by a codec. The following sections discuss the conversion of analog to digital.

Digitizing Analog Signals There are four steps in the process of digitizing analog sound:

1. Sample the analog sound at regular intervals

2. Quantize the sample

3. Encode the value into a binary expression

4. Optionally compress the sample

Sampling could be done any number of times per second; the more samples taken per second, the higher the audio

quality, but the amount of digital data produced is much larger. Nyquist's theorem states that the sampling interval should

be 2x the highest frequency of the sample to produce acceptable audio quality during playback. Because the highest

frequency in human speech that we want to reproduce in telephony is around 4000 Hz, the sampling rate for standard toll-

quality digital voice is 8000 intervals per second. By contrast, CD music audio, which must encode both much higher and

much lower frequencies, samples at about 192,000 times per second.

Quantizing refers to making a digital approximation of an analog waveform. Imagine drawing an arc on a chessboard; if

you had to define the arc using only the square it was in for each row (segment) and column (interval), you would end up

with a stepped pattern that was sort of close to the original arc but not exact. This is exactly the process that happens with

quantization: the codec chooses a segment value that is as close as possible to the analog value at the interval it was

sampled, but it cannot be exact. To make the quantization more accurate, each sample is divided into 16 intervals that are

adjusted to more closely match the sampled wave. Furthermore, the segments are actually more fine-grained at the origin

than at the high and low ranges. This is because most of the human speech we are trying to capture accurately is in this

center range of the scale; there are fewer sounds at the very highest and lowest values.


F IGURE 11 Quantizing the Digital Sample

Encoding the signal is a simple process. We have a single 8-bit code word to identify whether the analog signal was a

positive or negative voltage, what value the signal was quantized to (which segment), and finally, which interval is repre

sented by the code word. The first bit identifies either positive voltage (1) or negative (0). The next three bits represent

the segment. There are eight segments in the positive range and eight segments on the negative range, so three bits

provide the necessary encoding for the quantization. The last four bits identify the interval. A code word example is

shown next:

In this case, the first 1 indicates a positive voltage; the next digits of 001 indicate this is the first segment (on the positive

side), and 1100 indicates the twelfth interval.

The code word is 8 bits; we generate a code word 8000 times per second (the sample rate). This gives us a bitrate output

of 8 x 8000 = 64,000 bps (64 kbps). The process we just described is known as Pulse Code Modulation (PCM) and is the

standard for uncompressed digital voice in telephony. One voice stream thus requires 64k of bandwidth for transport.

1 0 0 1 1 1 0 0


NOTE The determination of voice quality is based on the Mean Opinion Score (MOS). This is a subjective measurement, created by gathering the opinion of live human listeners. A sample recording is played, and the listeners give it a score out of 5, where 5 is best. The same sample is played using different compression or processing methods and scored again. Because MOS is so subjective, other quality measurements exist that are more empirical and more accurate. For reference, standard PCM encoding (G.711) scores 4.1, and G.729 scores 3.92.

Compression is not a required step, but it is often done to save bandwidth in VoIP environments. The two main types of compression we are concerned with are the following:

• Adaptive Differential PCM (ADPCM): This method does not send entire code words, but instead sends a smaller

code that represents the difference between this word and the last one sent. This is not commonly used today,

because it produces lower voice quality and compresses down only to about 16 kbps.

• Conjugate Structure Algebraic Code Excited Linear Prediction (CS_ACELP): As the name suggests, this is

more complex compression. Based on a dictionary or codebook of known sounds made by a standardized American

male voice, the digital sample is analyzed and compared to the dictionary. The dictionary code that is the closest to

the sample is sent. The codebook is constantly learning. The output of this compression is typically 8 kbps—with

very little degradation of voice quality. This compression is widely used in VoIP.

Time Division Multiplexing (TDM) TDM is the primary technology used in traditional digital voice; it is also extensively used in data circuits. The basic

premise is to take pieces of multiple streams of digital data and interleave them on a single transmission medium.

T1 Circuits On a Tl circuit, there are up to 24 channels available for voice. 64k from conversation 1 is loaded into the first Tl

channel, then 64k from the conversation 2 is loaded into the second channel, and so on. If not enough conversations exist

to fill the available channels, they are padded with null values. The 24 channels are grouped together as a frame.

Depending on the implementation, either 12 frames are grouped together as a larger frame (called SuperFrame or SF), or



24 frames are grouped together (called Extended SuperFrame or ESF). T l s are typically full duplex, with two wires sending and the other two wires receiving.

E1 Circuits An El is very similar to a T l . There are 32 channels, of which 30 can be used for voice. (The other two are used for

framing and signaling, respectively.) The 32 channels are grouped together as a frame, and 16 frames are grouped

together as a multiframe. El circuits are common in Europe and Mexico, with some El services becoming available in

the United States.

Channel Associated Signaling (CAS)—T1 Although the 64 k channels of a Tl are intended to carry digitized voice, we must also be able to transmit signaling infor

mation, such as on-hook and off-hook, addressing, and so forth. In CAS circuits, the least significant bit of each channel

in every sixth frame is "stolen" to generate signaling bit strings. SF implementation takes 12 frames and creates a

SuperFrame. Using one bit per channel in every sixth frame gives two 12-bit signaling strings (known as A and B) per

SuperFrame. The A and B strings are used to signal basic status, addressing, and supervisory messages. In ESF, 24 chan

nels are in an Extended SuperFrame, which gives A, B, C, and D signaling strings. These can be used to signal more

advanced supervisory functions.

Because CAS takes one bit from each channel in every sixth frame, it is known as Robbed Bit Signaling (RBS). Using

RBS means that a slight degradation occurs in voice quality because every sixth frame has only 7 instead of 8 bits to

represent the sample; however, this is not generally a perceptible degradation.


Channel Associated Signaling (CAS)—T1 El signaling is slightly different. In an El CAS circuit, the first channel (channel 0 or timeslot 1) is reserved for framing

information. The 17th channel (channel 16 or timeslot 17) contains signaling information—no bits are robbed from the

individual channels. Timeslots 2-16 and 18-32 carry the voice data. Each channel has specific bits in timeslot 17 for

signaling. This means that although El CAS does not use RBS, it is still considered CAS; however, the signaling is out-

of-band in its own timeslot.

Common Channel Signaling (CCS) CCS provides for a completely out-of-band signaling channel. This is the function of the D channel in an ISDN PRI or

BRI implementation. The full 64 k of bandwidth per channel is available for voice; instead of generating ABCD bits, a

protocol known as Q.931 is used out-of-band in a separate channel for signaling. An ISDN PRI Tl provides 23 voice

channels of 64 k each (called Bearer or B channels) and one 64 k D (for Data) channel (timeslot 24) for signaling. An

ISDN PRI El provides 30 B channels and 1 D channel (timeslot 17); an ISDN BRI circuit provides two 64 k B channels

and one D channel of 16 k.

Understanding Packetization IP networks move data around in small pieces known as packets. Because we know how to digitize our voice, it now

becomes just another binary payload to move around in a packet. VoIP uses Digital Signal Processors (DSP) for the codec

functions. The digitized voice is then packaged in an appropriate protocol structure to move it through the IP infrastructure.

DSPs DSPs are specialized chips that perform high-speed codec functions. DSPs are found in the IP phones to encode the

analog speech of the user and to decode the digitized contents of the packets arriving from the other end of the call. DSPs

are also used on IOS gateways at the interface to PSTN circuits, to change from a digital circuit to packetized voice, or

from an analog circuit to packetized voice. DSPs also change from one codec to another, allow conferencing and call

park, and other telephony features. DSPs are a vital component of a VoIP system. Different chip types have varying

capacities, but the general rule is that you want as many DSP resources available to you as possible. The DSP calculator

on cisco.com will help you calculate what you must have.

Real-Time Transport Protocol (RTP) RTP was developed to better serve real-time traffic such as voice and video. Voice payloads are encapsulated by RTP, then

by UDP, then by IP. A Layer 2 header of the correct format is applied; the type obviously depends on the link technology

in use by each router interface. A single voice call generates two one-way RTP/UDP/IP packet streams. UDP provides

multiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering.


Understanding VoIP The elements of traditional telephony—status, address and supervisory signaling, digitization, and so on—must have

functional parallels in the VoIP world for systems to function as people expect them to, and more importantly, for VoIP to

interact with the PSTN properly.

This section examines packetizing digital voice, signaling, and transport protocols, the components of a VoIP network,

and the factors that can cause problems in VoIP networks and how they can be mitigated.

http://cisco.com


Payload identification allows us to treat voice traffic differently from video, for example, simply by looking for the RTP

header label, simplifying our configuration tasks. Timestamping and sequence numbering allows VoIP devices to reorder

RTP packets that arrived out of sequence and play them back with the same timing in which they were recorded, elimi

nating delays or jerkiness. There is no provision for retransmission of a lost RTP packet.

Each RTP stream is accompanied by a Real-Time Transport Control Protocol (RTCP) stream. RTCP monitors the quality

of the RTP stream, allowing devices to record events such as packet count, delay, loss, and jitter (delay variation).

A single voice packet by default contains a payload of 20 msec of voice (either uncompressed or compressed). Because

sampling is occurring at 8000 times per second, 20 msec gives us 160 samples. If we divide 8000 by 160, we see that we

are generating 50 packets with 160 bytes of payload, per second, for a one-way voice stream.

If we use compression, we can squeeze the 160-byte payload down to 20 bytes using the G.729 codec. We still have 160

samples, still 20 msec of audio, but reduced payload size.

Codecs The codecs supported by Cisco include the following:

• G.711 (64kbps)—Toll-quality voice, uncompressed.

• G.729 (8kbps)

• Annex A variant: less processor-intensive, allows more voice channels encoded per DSP chip; lower audio

quality than G.729

• Annex B variant: Allows the use of Voice Activity Detection and Comfort Noise Generation; can be applied to

G.729 or G.729-A

The values for bandwidth shown do not include the Layer 3 and Layer 2 overhead; the actual bandwidth used by a single

(one-way) voice stream can be significantly larger. The following tables summarize the additional overhead added by

packetization and Layer 2 encapsulation (assume 50 packets per second (pps):

Bandwidth Calculation, Without Layer 2

Codec G.711 G.729

Voice Payload 160 Bytes 20 Bytes

RTP Header 12 Bytes 12 Bytes

UDP Header 8 Bytes 8 Bytes

IP Header 20 Bytes 20 Bytes

Total Before Layer 2 200 Bytes 60 Bytes

Total Bitrate @ 50 pps 80,000 bps (80 kbps) 24,000 bps (24 kbps)

Bandwidth Calculation, With Layer 2

Layer 2 Type G.711 = 200 Bytes/packet G.729 = 60 Bytes/packet

Ethernet 18 Bytes 18 Bytes

Multilink PPP 6 Bytes 6 Bytes

Frame Relay FRF. 12 6 Bytes 6 Bytes

Total incl. Layer 2 218 Bytes 206 Bytes 206 Bytes 78 Bytes 66 Bytes 66 Bytes

Total Bitrate incl. Layer 2 (@ 50 pps)

87.200 82,400 82,400 (87.2 kbps) (82.4 kbps) (82.4 kbps)

31,200 26,400 26,400 (31.2 kbps) (26.4 kbps) (26.4 kbps)

When using G.729, the RTP/UDP/IP header of 40 bytes is twice the size of the 20B voice payload. This consumes signif

icant bandwidth just for header transmission on a slow link. The recommended solution is to use Compressed RTP

(cRTP) on slow WAN links. cRTP reduces the RTP/UDP/IP header to 2 bytes without checksums or 4 bytes with check

sums. The effect of using cRTP is illustrated in the following table. (Note: Ethernet is not included because it is not clas

sified as a slow link.)


Bandwidth Calculation, Using cRTP

Codec G.711 G.729

Voice Payload 160 Bytes 20 Bytes

cRTP header w/ chksum 4 Bytes 4 Bytes

cRTP header no chksum 2 Bytes 2 Bytes

Total before Layer 2: 164 Bytes 162 Bytes 24 Bytes 22 Bytes

Multilink PPP or

Frame Relay FRF. 12 6 Bytes 6 Bytes

Total WAN bandwidth @50 pps incl. Layer 2: 68000 bps 67,200 bps (68 kbps) (67.2 kbps)

12,000 bps 11,200 (12 kbps) (11.2 kbps)

Voice Activity Detection (VAD) Phone conversations on average include about 35% silence. In Cisco Unified Communications, by default silence is pack

etized and transmitted, consuming the same bandwidth as speech. In situations where bandwidth is very scarce, the VAD

feature can be enabled, causing the voice stream to be stopped during periods of silence. The theory here is that the band

width otherwise used for silence can be reclaimed for voice or data transmission. VAD also adds Comfort Noise

Generation (CNG), which fills in the dead silence created by the stopped voice flow with white noise. VAD should not be

taken into account during the network design bandwidth allocation process because its effectiveness varies with back

ground noise and speech patterns. VAD is also made ineffective by Music on Hold and fax features. In reality, VAD typi

cally causes more problems than it solves, and it is usually wiser to add the necessary bandwidth.



Additional DSP Functions In addition to digitizing voice, DSP resources are used for the following:

• Conferencing: DSPs mix the audio streams from the conference participants and transmit the mix (minus their own)

to each participant.

• Transcoding and Media Termination Points (MTP): A transcoder changes a packetized audio stream from one

codec to another, perhaps for transit across a slow WAN link. MTPs provide a point for the stream to be terminated

while other services are set up.

• Echo Cancellation: DSPs provide the calculation power needed to analyze the audio stream and filter out the repeti

tive patterns that indicate echo. Echo is a chief cause of perceived poor voice quality; echo cancellation is an impor

tant function.

H.323 Yes--ITU Very Good Yes No Peer-to-Peer

MGCP Yes--IETF Good Yes No Client/Server

SIP Yes--IETF Basic Yes Yes; also third-party phones Peer-to-Peer

SCCP N o - -Cisco Proprietary Cisco only Some Cisco IP Phones only Client/Server

H.323 H.323 is not itself a protocol; it is an umbrella standard that defines several other related protocols for specific tasks.

Originally conceived as a multimedia signaling protocol to emulate traditional telephony functionality in IP LAN

Introducing VoIP Signaling Protocols VoIP signaling protocols are responsible for call setup, maintenance, and teardown. A number of different protocols are in

use—some standards-based, others proprietary, and each with advantages and disadvantages. The following sections

introduce the signaling protocols you should know about, including SCCP, H.323, MGCP, and SIP.

VoIP Signaling Protocols VoIP signaling protocols handle the call setup, maintenance, and teardown functions of VoIP calls. It is important to keep

in mind that the signaling functions are an entirely separate packet stream from the actual voice bearer path (RTP). The

signaling protocol in use must pass the supervisory, informational, and address information expected in any telephony

system.

VoIP signaling protocols are either peer-to-peer or client-server; in the case of peer-to-peer protocols, the endpoints have

the intelligence to perform the call-control signaling themselves, Client-server protocols send event notifications to the

call agent (the Unified CM server) and receive instructions on what actions to perform in response. The following table

summarizes the characteristics of the four signaling protocols dealt with here.

Inter-Vendor Implemented Implemented Protocol Standard? Compatibility on Gateways on Cisco IP Phones Operating Mode



environments, it is a long-established and stable protocol very suitable for intervendor compatibility. H.323 is supported

by all Cisco voice gateways and CM platforms as well as some third-party video endpoints.

MGCP Media Gateway Control Protocol is a lightweight client/server protocol for PSTN gateways and some clients. It is simple

to configure and allows the call agent to control the MGCP gateway, eliminating the need for expensive gateways with

intelligence and complex configurations. The gateway reports events such as a trunk going off-hook, and the call agent

instructs the gateway on what to do; the gateway has no local dial plan because all call routing decisions are made at the

call agent and relayed to the MGCP gateway. MGCP is not as widely implemented as SIP or H.323. MGCP is not

supported by Unified CM Express or the Smart Business Communication System.

SIP Session Initiation Protocol is an IETF standard that uses peer-to-peer signaling. It is very similar in structure and syntax

to HTTP, and because it is text-based, it is relatively simple to debug and troubleshoot. SIP can use multiple transport

layer protocols and can support security and proxy functions. SIP is an evolving standard that currently provides basic

telephony functionality; further developments and extensions to the standard will soon make it feature-comparable with

SCCP. One of its most important capabilities is creating SIP trunks to IP Telephony service providers, replacing or

enhancing traditional TDM PSTN connections.

SCCP Skinny Client Control Protocol is a Cisco-proprietary signaling protocol used in a client-server manner between Unified

CM and Cisco IP Phones (and some Cisco gateways). SCCP uses TCP connections to the Unified CM to set up, maintain,

and tear down voice and video calls. It is referred to as a stimulus protocol, meaning that it sends messages in response to

events such as a phone going off-hook or a digit being dialed. SCCP is the default signaling protocol for all Cisco IP

phones, although many also support SIP; SIP does not yet support the full feature set available to SCCP phones. All

Cisco Unified Communications call agents (CM, CM Express, and the 500 Series) and some gateways support SCCP.


Connect ing a VoIP System to a Serv ice Provider Ne twork A VoIP system that can place calls only to other devices on the IP network is only marginally useful; we still need to

place calls out to the PSTN, and to do so we need to connect to a phone service provider, whether via traditional TDM

links or ITSP connections. The device that acts as the interface to the PSTN is the voice gateway; it provides the physical

connection and logical translation between two or more different network technologies.

Understanding Gateways, Voice Ports, and Dial Peers The following sections establish some terms of reference.

Gateways In the Cisco Unified Communications architecture, a gateway is typically a voice-enabled router with an appropriate

voice port installed and configured. Gateways can have both analog and digital voice port connections, including analog

FXO, FXS, and E&M or digital T l / E l or PRI interfaces.

Call Legs A call leg is the inbound or outbound call path as it passes through the gateway. As the call comes into the gateway, it is

associated with an inbound port. (This is the inbound call leg.) As the call is sent out another gateway port, this creates

the outbound call leg. There will be inbound and outbound call legs at each gateway router.

Dial Peers A dial peer is a pointer to an endpoint, identified by an address (a pattern of digits). Cisco gateways support two types of

dial peers: POTS and VoIP. POTS dial peers are addressed with PSTN phone numbers, and VoIP dial peers are addressed

by IP addresses. Dial peers identify the source and destination endpoints of call legs; an inbound call leg is matched to a

dial peer, and the outbound call leg is routed to a destination dial peer. Depending on the direction of the call, the dial peers

may be POTS inbound and VoIP outbound, vice versa, or possibly both VoIP. It is unlikely but not impossible that the

+ The preceding digit is repeated one or more times.

* and # NOT wildcards; these are valid DTMF digits.

, (comma) Inserts a one-second pause.

. (dot) Specifies any one wildcard digit (0 - 9, *, #). The pattern "20." would match all strings from 200 through 209, plus 20*and20#.

[ ] Square brackets define a range, within which any one digit may be matched; for example, "20[4-6]" defines 204, 205, and 206.

T Indicates a variable length string; this is useful in cases where local, long-distance, and international PSTN numbers may be called; the destination pattern could men be ".T". This pattern would match any string of up to 32 digits.


inbound and outbound dial peers would both be POTS. Each dial peer also defines attributes such as the codec to use, QoS

settings, and other feature settings. Dial peers are configured in the gateway IOS, using either the CLI or GUI interface.

The partial output that follows shows a simple POTS dial peer configuration:

G a t e w a y ( c o n f i g ) # d i a l - p e e r v o i c e 10 pots

G a t e w a y ( c o n f i g - d i a l p e e r ) # d e s t i n a t i o n - p a t t e r n 8675309

G a t e w a y ( c o n f i g - d i a l p e e r ) # p o r t 1 /0/1

The number assigned to dial peers is arbitrary, although dial peer 0 exists by default and cannot be deleted. The keyword

pots creates a POTS dial peer; the keyword voip would create a VoIP dial peer. The destination-pattern command iden

tifies that the attached device (phone or PBX) terminates calls to the specified number (or a range of numbers if connect

ing to a PBX). The port command identifies the physical hardware connection the dial peer will use to reach the

destination pattern.

The destination-pattern command associates a phone number with a dial peer. The specified pattern can be a specific

phone number (as above, 8675309) or an expression that defines a range of numbers. The router uses the patterns to

decide which dial peer (and associated physical port) it should route a call to. The following table briefly explains

destination-pattern syntax.

Character Meaning

C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e Prov ider N e t w o r k Configuring VoIP dial peers is equally simple. Examine the following configuration:

G a t e w a y ( c o n f i g ) # d i a l - p e e r vo i ce 20 v o i p

G a t e w a y ( c o n f i g - d i a l p e e r ) # d e s t i n a t i o n - p a t t e r n 4 . . . .

G a t e w a y ( c o n f i g - d i a l p e e r ) # sess ion t a r g e t i p v 4 : 1 0 . 1 . 1 . 2

In this example, the destination pattern is any four-digit number starting with "4." A new command, session-target, is

used to identify the IP (version 4 in this case) address of the gateway or call agent that will terminate the call. If the IP

address is on a router, it should be a loopback IP so that the address is always available even if a physical interface fails.

The preceding configuration creates an H.323 dial peer (in contrast to a SIP dial-peer).

NOTE The default dial peer 0 cannot be deleted or modified. It does not negotiate services such as VAD, DTMF Relay, or TCL applications. The dial peer 0 configuration for inbound VoIP calls contains the following:

• Any codec

• VAD enabled

• No RSVP Support

• Fax-rate voice

Routers attempt to match dial peers for the inbound call leg according to the following rules:

1. Look for the incoming called-number command in a dial peer that matches the called number or DNIS string of the

inbound leg.

2. Look for the answer-address command in a dial peer that matches the calling number or ANI string of the inbound

call leg.

3. Look for the destination-pattern command in a dial peer that matches the calling number or ANI string of the

inbound call leg.

4. Look for the POTS dial peer port command that matches the voice port of the incoming call (POTS dial peers only).

5. If all of the above fail to match, match against Default Dial Peer 0 as a last resort.

The default dial peer 0 config for inbound POTS calls includes the following:

• no ivr application

When a router is matching the dialed digits against the patterns in the configured dial peers, it attempts to find the longest

match. This occurs on a digit-by-digit basis. Each successive digit may validate some patterns while eliminating others

until a single pattern represents the longest match between the dialed digits and the destination pattern, at which point the

call is routed to the outbound dial peer configured with that matching pattern.


Internet Telephony Service Providers As VoIP technology matured and stabilized, telephone service providers began extending VoIP connectivity to their

customers, allowing for simple, flexible connection alternatives to traditional TDM links. Internet Telephony Service


Consider the following configuration:

d i a l - p e e r v o i c e 10 v o i p

d e s t i n a t i o n - p a t t e r n .T

sess ion t a r g e t i p v 4 : 1 0 . 1 0 . 1 0 . 1

i

d i a l peer v o i c e 20 v o i p

d e s t i n a t i o n - p a t t e r n 8 6 7 [ 2 - 3 ] . . .


!


d e s t i n a t i o n - p a t t e r n 8 6 7 4 . . .

sess ion t a r g e t i p v 4 : 10 .10 .30 . 1

i


d e s t i n a t i o n - p a t t e r n 8675309


Given this configuration, the following example dialed numbers illustrate how the patterns match dialed digits:

• The dialed number 867-5309 will match dial peer 40 (exact 7-digit match)

• The dialed number 867-4309 will match dial peer 30 (first four digits match)

• The dialed number 867-3309 will match dial peer 20 (first four digits match)

• The dialed number 876-5309 will match dial peer 10 (no other exact match, so the " .T" pattern matches)


Providers (ITSP) connections are typically much less expensive, available in smaller bandwidth increments than Tl or

PRI links, and can route nonvoice data traffic concurrently. QoS configuration is supported (and in fact is required for

proper VoIP operation). Most ITSP links use SIP, but H.323 is an option. The gateway configuration is relatively simple,

with the creation of a VoIP dial peer pointing at the provider with the parameters they supply. PSTN calls are routed to

the provider, who then routes calls to their PSTN connection, usually with a toll-minimizing route that dramatically

reduces long-distance costs to the customer.

Understanding Call Setup and Digit Manipulation Successfully completing a phone call requires that the correct digits are sent to the terminating device, whether on the

VoIP network or the PSTN. PSTN calls are typically more complex because of the varying local and international

requirements for the number of digits required to route the call. Over and above this basic requirement are the additional

complexities imposed by requirements of the business: we may want to change our ANI number, add or strip access

codes, compensate for undesirable default behavior, or build specialized functionality for our particular purposes. This

section deals with digit manipulation and troubleshooting.

Digit Consumption and Forwarding Some strange things happen when an IOS gateway matches a dial peer for an outbound call leg and forwards the dialed

digits to the terminating device.

For POTS dial peers, the gateway consumes (meaning strips away) the left-justified digits that exactly match the dial-peer

destination pattern and forwards only the wildcard-matched digits to the terminating device. Clearly, this could cause

problems if the PSTN were to receive only 4 digits, as in this example:

d i a l - p e e r v o i c e 20 pots

d e s t i n a t i o n - p a t t e r n 8 6 7 . . . .

po r t 1/0:1


With this configuration, if the dialed number was 867-5309, the gateway would forward only 5309 (the wildcard

matches), and the PSTN would be unable to route the call. Adding the command no digit-strip in the dial-peer configura

tion will change this behavior and cause the gateway to forward all dialed digits.

For VoIP dial peers, the default behavior is to forward all collected digits.

Digit Collection The router will collect digits one at a time and attempt to match a destination pattern. As soon as it has an exact match,

the call is immediately placed, and no more digits are collected. If there are destination patterns that have overlapping

digits, this can cause calls to be misrouted, as in the following example:

D i a l - p e e r v o i c e 1 v o i p

D e s t i n a t i o n p a t t e r n 555

Session t a r g e t i p v 4 : 1 0 . 1 . 1 . 1

!


D e s t i n a t i o n - p a t t e r n 5552112


If the user dials 555-2112, dial peer 1 will exactly match at the third digit, the call will be immediately routed using dial

peer 1, and only the collected digits of 555 will be forwarded. We solve the problem by changing the configuration as

shown next:


D e s t i n a t i o n p a t t e r n 5 5 5 . . . .


!


D e s t i n a t i o n - p a t t e r n 5552112



Now, when the third digit is entered, the router cannot make an exact match because both dial peers are possible matches;

when the last digit is dialed, the router determines that dial peer 2 is an exact match and immediately places the call. Dial

peer 1 is also a match, but because of the wildcards, the destination pattern matches 10,000 possible numbers (0000

through 9999); it is not as close a match as dial peer 2.

Digit Manipulation Sometimes we need to add, change, or remove digits before the call is placed. We do this to avoid inconveniencing users

or to match the dialed digit requirements of a gateway or the PSTN. We have several methods of modifying the digit

string, as described in the following sections.

prefix

The prefix dial-peer command adds digits to the beginning of the string after the outbound dial peer is matched but

before passing digits to the destination. An example of its use is a POTS dial peer with 2 . . . as the destination pattern. If

the user dials 2112, the default behavior is for the POTS dial peer to forward only 112. Adding the command prefix

6045552 forces the router to prepend the additional digits required to route the call over the PSTN:

d i a l - p e e r v o i c e 20 po ts

d e s t i n a t i o n - p a t t e r n 2 . . .

p r e f i x 6045552

p o r t 1 /0 /0

forward-digits

forward-digits: This dial-peer command forces the specified number of digits to be forwarded, whether the digits were

exact match or wildcard matches, overriding the default behavior of stripping the exact matches. You can specify a

number of digits to forward (as shown in the example that follows) or use forward-digits all to force all dialed digits to

be forwarded.




f o r w a r d - d i g i t s 7

p o r t 1 /0 /0

Number Expansion

num-exp: The number expansion table is a global command that either expands an extension (perhaps a 4-digit extension

into a full 10-digit PSTN number) or completely replaces one number with another. This command is applied before the

outbound dial peer is matched, so there must be a configured dial peer that matches the expanded number for the call to

be forwarded.

num-exp 2 . . . 5 5 5 2 . . .



p o r t 1 /0 /0

Translation Rules

voice translation-rule: This global command configures number translation profiles to allow us to alter the ANI, DNIS,

or redirect number for a call. Using the command is a three-step process:

1. Define the translation rule globally:

vo i ce t r a n s l a t i o n - r u l e 1

r u l e 1 /555 / / 867 /

The rule command defines a pattern to match (in this case 555) and a pattern to change the matched digits to (in this

case 867). The match and replace patterns are identified and separated by the "/" characters that begin and end the

patterns.

Cisco Regular Expression Characters for Voice Translation Rules

Character Description

Matches any single character.

\ (match) In the match phrase: Escape the special meaning of the next character.

\(replace) In the replace phrase: Reference a set number from the match phrase.

A Match the expression at the beginning of the digit string.

$ Match the expression at the end of the digit string.

/ Identifies the start and end of both the match and replace phrases.

[0-9] Match a single character in a list.

[A0-9] Do not match a single character specified in the list.

continues


2. Create the voice translation profile containing the translate instruction (the options are [calledlcallingl redirect-calledlredirect-target], and reference the rule we just defined by number. In this example we are translat

ing the called number:

vo i ce t r a n s l a t i o n - p r o f i l e JENNY

t r a n s l a t e c a l l e d 1

3. Apply the profile to one or more dial peers, either inbound or outbound:

d i a l - p e e r vo i ce 20 po ts

d e s c r i p t i o n t r a n s l a t e d t o Jenny

t r a n s l a t i o n - p r o f i l e outgoing JENNY


p o r t 1 /0 /0

Translation rules use regular expression syntax, which can be quite complex. The following table defines the characters

used, and examples follow.

* Repeat the previous expression 0 or more times.

+ Repeat the previous expression 1 or more times.

? Repeat the previous expression 0 or 1 time.

( ) Identifies a set in the match expression.

Example 1:

r u l e 1 /123 / / 456 /

The first set of forward slashes defines the match phrase; the second set defines the replace phrase. This expression means

"match 123 and replace it with 456." Thus:

• 123 is replaced with 456



• 1234123 is replaced with 4564123 (only the first instance of the match is replaced)

Example 2:

vo i ce t r a n s l a t i o n ? r u l e 1

r u l e 1 / A 4 0 . . . / /66660B0/

This example replaces any five-digit number that begins with "40" with the number "6666000".

Example 3:

vo i ce t r a n s l a t i o n ? r u l e 1

/ A \ ( 8 6 7 \ ) \ ( . . . . \ ) / / 5 5 5 \ 2 /


Cisco Regular Expression Characters for Voice Translation Rules continued


This example means: "If the number starts with 867 and is followed by any four other digits, change the 867 to 555 and

replace the other four digits with the digits in Set 2 of the match." Remember that the forward slashes define the match

and replace phrases; the backslashes mean "the next character is not part of what to match"; the round brackets indicate

which sets of characters in the matched number to keep in the replaced number. The sets are numbered starting with 1, so

the first set of round brackets is 1, and the second is 2 (as in this example).

Private Line Automatic Ringdown (PLAR) PLAR creates a permanent association between a voice port and a destination number (or voice port). When PLAR is

configured, going off-hook on that voice port automatically dials the pattern specified by the connection plar <number>

command. The caller does not hear a dial tone and does not have to dial a number. Think of PLAR as a hotline; pick up

the Batline and you get Batman without having to dial.

The following shows a simple PLAR configuration that will call 867-5309 when the phone goes off-hook:

vo i ce p o r t 1 /0 /0

connec t ion p l a r 8675309

Troubleshooting Dial Plans and Dial Peers The following sections discuss some of the commands available to troubleshoot your configuration.

show dial-peer voice To display information for voice dial peers, use the show dial-peer voice command in user EXEC or privileged EXEC

mode.

show d i a l - p e e r vo i ce [number | summary]

Syntax Description

number (Optional) A specific voice dial peer. Output displays detailed information about that dial peer.

summary (Optional) Output displays a short summary of each voice dial peer.

If both the name argument and summary keyword are omitted, output displays detailed information about all voice dial

peers.

The following is sample output from this command for a VoIP dial peer:

Router# show d i a l - p e e r v o i c e 101

VoiceOverIpPeer101

peer t ype = v o i c e , i n f o r m a t i o n t ype = v o i c e ,

d e s c r i p t i o n = ' ' ,

tag = 6001 , d e s t i n a t i o n - p a t t e r n = " 6 0 0 1 ' ,

answer-address = ' ' , p re fe rence=0 ,

CLID R e s t r i c t i o n = None

CLID Network Number = " 1

CLID Second Number sent

CLID O v e r r i d e RDNIS = d i s a b l e d ,

source c a r r i e r - i d = t a r g e t c a r r i e r - i d =

source t r u n k - g r o u p - l a b e l = ' ' , t a r g e t t r u n k - g r o u p - l a b e l = ' ' ,

numbering Type = "unknown'

group = 6001 , Admin s t a t e i s up, Opera t i on s t a t e i s up,

incoming c a l l e d - number = connect ions/maximum = 0 / u n l i m i t e d ,

DTMF Relay = d i s a b l e d ,

<outpu t om i t t ed>


d i a l - p e e r hunt 0

PASS

TAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF THRU SESS TARGET

100 pots up up 0

101 vo ip up up 5550112 0 sys t i p v 4 10 .10 .1 1

102 vo ip up up 5550134 0 sys t i p v 4 10 .10 .1 1

99 vo ip up down 0 sys t

33 pots up down 0

debug voip dialpeer inout

To display information about the voice dial peers, use the debug voip dialpeer command in privileged EXEC mode.

Router# debug v o i p d i a l p e e r i n o u t

v o i p d i a l p e e r i n o u t debugging i s on

*May 1 1 9 : 3 2 : 1 1 . 7 3 1 : / / -1 /6372E2598012/DPM/dpAssoc ia te IncomingPeerCore:

Resu l t=Success(0) a f t e r DP_MATCH_INCOMING_DNIS; Incoming Dia l -peer=100


C a l l i n g Number=4085550111, C a l l e d Number=3600, V o i c e - l n t e r f a c e = 0 x 0 ,

Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

Peer I n f o Type=DIALPEER_INFO_SPEECH


Resu l t=Success(0) a f t e r DP_MATCH_INCOMING_DNIS; Incoming D ia l - pee r=100

*May 1 1 9 : 3 2 : 1 1 . 7 3 5 : / / -1/6372E2598012/DPM/dpMatchPeersCore:

C a l l i n g Number=, C a l l e d Number=3600, Peer I n f o Type=DIALPEER_INFO_SPEECH


The following is sample output from this command with the summary keyword:

Router# show d i a l - p e e r v o i c e summary

Troubleshooting Signaling for POTS Call Legs

show controllers t1 The show controllers tl command displays Tl (or E l ) controller status and function. The following is sample output

from this command:

Router# show c o n t r o l l e r s t1

T1 4 /1 i s up.

App l i que t ype is Channe l ized T1

Cab le leng th i s s h o r t 133

No alarms d e t e c t e d .

Framing is ESF, L ine Code is AMI, Clock Source is l i n e

Data i n c u r r e n t i n t e r v a l (10 seconds e l a p s e d ) :

0 L ine Code V i o l a t i o n s , 0 Path Code V i o l a t i o n s 0 S l i p Sees, 0 Fr Loss Sees,


*May 1 19 :32 :11 .735 : / / -1/6372E2598012/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; C a l l e d Number=3600

*May 1 1 9 : 3 2 : 1 1 . 7 3 5 : / / -1/6372E2598012/DPM/dpMatchPeersCore:

Resu l t=Success(0) a f t e r DP_MATCH_DEST

*May 1 1 9 : 3 2 : 1 1 . 7 3 5 : / / -1/6372E2598012/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

The following event shows the matched dial peers in the order of priority:

List of Matched Outgoing Dial Peer(s):

1: Dial PeerTag=3600

2: Dial Peer Tag=36

C o n n e c t i n g a VoIP S y s t e m to a S e r v i c e Prov ider N e t w o r k 0 L ine E r r Sees, 0 Degraded Mins 0 E r ro red Sees, 0 Bu rs t y E r r Sees,

0 Severe ly E r r Sees, 0 U n a v a i l Sees

In this output, no alarms were detected. Possible alarms are as follows:

• Transmitter is sending remote alarm.

• Transmitter is sending AIS.

• Receiver has loss of signal.

• Receiver is getting AIS.

• Receiver has loss of frame.

• Receiver has remote alarm.

• Receiver has no alarms.

show voice port Use the show voice port command to display configuration and voice-interface-card-specific information about a specific

port.

The following is sample output for an E&M analog voice port:

Router# show v o i c e p o r t 1 /0 /0

E&M S l o t i s 1 , S u b - u n i t i s 0 , Po r t i s 0

Type of Vo icePor t is E&M

Opera t ion S t a t e is DORMANT

A d m i n i s t r a t i v e S t a t e is UP

I n i t i a l Time Out is se t to 0 s

I n t e r d i g i t Time Out is se t to 0 s



Analog I n f o F o l l o w s :

Region Tone i s se t f o r no r thamer i ca

Voice card s p e c i f i c I n f o F o l l o w s :

S i g n a l Type i s w i n k - s t a r t

Opera t ion Type i s 2 - w i r e

E&M Type is 1

D i a l Type i s dtmf

I n Se izu re i s i n a c t i v e

Out Se izu re i s i n a c t i v e

show dialplan number To display which outgoing dial peer is reached when a particular telephone number is dialed, use the show dialplan

number command in privileged EXEC mode.

Router# show d i a l p l a n number 1001

Macro Exp . : 1001

VoiceEncapPeer1003

i n f o r m a t i o n type = v o i c e ,

t a g = 1003, d e s t i n a t i o n - p a t t e r n = 1 0 0 1 ' ,

answer-address = ' ' , p re fe rence=0 ,

numbering Type = 1 unknown'

group = 1003, Admin s t a t e is up, Opera t i on s t a t e is up,

incoming ca l l ed -number = ' ' , connect ions/maximum = 0 / u n l i m i t e d ,

DTMF Relay = d i s a b l e d ,

hun ts top = enab led ,

t ype = p o t s , p r e f i x = ' ' ,

f o r w a r d - d i g i t s d e f a u l t

s e s s i o n - t a r g e t = 1 ' , vo ice-port = '1/1'

debug voip ccapi inout The debug voip ccapi inout command traces the execution path through the call control API, which serves as the inter

face between the call session application and the underlying network-specific software. You can use the output from this

command to understand how calls are being handled by the router. This command shows how a call flows through the

system. Using this debug level, you can see the call setup and teardown operations performed on both the telephony and

network call legs.

Router# debug v o i p c c a p i i n o u t

v o i p c c a p i i n o u t debugging i s on

NOTE

This debug generates a

very long output, which

is impractical to fully

duplicate here. I suggest

you familiarize yourself

with sample outputs from

the Cisco IOS Debug

Command Guide or

better yet from your own

lab experimentation.

The following lines show information about the calling and called numbers. The network presentation indicator (NPI)

shows the type of transmission. The Incoming Dial-Peer field shows that the incoming dial peer has been matched.

*Apr 18 2 0 : 4 2 : 1 9 . 3 4 7 : / / -1 /9C5A9CA88009/CCAPI/cc_api_cal l_setup_ind_common:

In te r face=0x64F26F10, C a l l I n f o (

Calling Number=4085550111(TON=National , NPI=ISDN, Screening=User , Passed,

P r e s e n t a t i o n = A l l o w e d ) ,

Called Number=83103(TON=Unknown, NPI=Unknown),

C a l l i n g Translated=FALSE, Subs r i be r Type S t r = R e g u l a r L i n e , F ina lDes t i na t i onF lag=TRUE,

Incoming Dial-peer=1, Progress I n d i c a t i o n = N U L L ( 0 ) , C a l l i n g IE Present=TRUE,

Source T rkg rp Route Labe l= , Target T rkg rp Route Labe l= , CLID Transparent=FALSE), C a l l

I d= -1

*Apr 18 2 0 : 4 2 : 1 9 . 3 4 7 : / / -1 /9C5A9CA88009/CCAPI /ccCheckCl ipCl i r :

I n : C a l l i n g Number=4085550111(TON=National, NPI=ISDN, Screen ing=User , Passed,

P resen ta t i on=A l l owed)

*Apr 18 2 0 : 4 2 : 1 9 . 3 4 7 : / / -1 /9C5A9CA88009/CCAPI /ccCheckCl ipCl i r :

Out: C a l l i n g Number=4085550111(TON=National, NPI=ISDN, Screen ing=User , Passed,

P resen ta t i on=A l l owed)



Preparing the Infrastructure to Support Unif ied Communicat ions In this section, the best practices components for preparing the network to properly support Unified Communications are

explored. Topics covered include the following:

• Voice VLANs

• DHCP

• NTP

• Power over Ethernet

• IP Phone firmware and configuration files

Voice VLANs VLANs provide a logical separation of Layer 3 traffic and are created at Layer 2 (the network switch). A voice VLAN

(VVLAN, also called an Auxiliary VLAN) is an additional VLAN for the exclusive use of VoIP and video traffic. The

benefits of using a VVLAN include isolation from the broadcast traffic data VLANs, a measure of additional security, and

simpler deployment because you do not have to renumber the IP address scheme of the whole network to add VoIP

endpoints. (Each VLAN is a new, separate subnet.)

Most Cisco IP Phones are actually 3-port switches. The port that connects to the network switch can act as an 802. lq

trunk, allowing both voice and data traffic to be multiplexed in their respective VLANs on the single cable to the network

switch. The second port connects the desktop PC to the phone (and thus to the network over the trunk on the first port),

and the third port is an internal one for the voice traffic generated and received by the phone.

On many Cisco switches, the port connecting the phone does not need to be a trunk; it can be an access port instead. The

switch is capable of sending the VVLAN ID using CDP messages, and the phone then sends frames from itself tagged

with the learned VVLAN ID and forwards frames from the attached PC untagged. These untagged frames will be tagged

with the access VLAN ID configured on the switch port when they are processed by the switch.


The phone adds a QoS marking to its own frames, using the 802. lq frame header Class of Service (CoS) field. The phone

marks its frames as CoS 5 by default. This is the recommended setting, but it can be modified.

The following is a typical switchport configuration for an attached IP Phone in VVLAN 100 and the PC in VLAN 50:

S w i t c h ( c o n f i g ) # i n t e r f a c e Fas tE the rne t 0/1

S w i t c h ( c o n f i g - i f ) # s w i t c h p o r t mode access

S w i t c h ( c o n f i g - i f ) # s w i t c h p o r t access v l a n 50

S w i t c h ( c o n f i g - i f ) # s w i t c h p o r t vo i ce v l a n 100

S w i t c h ( c o n f i g - i f ) # s p a n n i n g - t r e e p o r t f a s t

DHCP It is recommended that you use DHCP for IP Phone addressing. Create a separate subnet for the Voice VLAN and add the

Option 150 parameter to identify the TFTP server IP address. This can be done on an existing DHCP server, or a new one

can be added if necessary; Cisco routers have DHCP server capability. The following configuration is a typical example

of router-based DHCP to support IP Phones:

s e r v i c e dhcp

! enables the DHCP s e r v i c e

I

i p dhcp exc luded-address 1 0 . 1 . 1 . 1 1 0 . 1 . 1 . 1 0

! s p e c i f i e s a s t a r t / e n d range of addresses t h a t DHCP w i l l NOT ass ign

ip dhcp poo l name IP_PH0NES

! Creates a poo l of addresses ( c a s e - s e n s i t i v e name) and en te r s DHCP c o n f i g u r a t i o n mode

I network 1 0 . 1 . 1 . 0 255 .255 .255 .0 ! De f ines the subnet o f addresses f o r t he poo l

d e f a u l t - r o u t e r address 1 0 . 1 . 1 . 1

! De f ines the d e f a u l t gateway

dns -se rve r address 192 .168 .1 .10 192 .168 .1 .11

! i d e n t i f i e s t h e DNS se rve r IP address(es) - up to 8 I P ' s

!

o p t i o n 150 ip 192 .168 .1 .2

! i d e n t i f i e s the TFTP se rve r IP

If you choose to use a DHCP server that resides on a different network, you will need to add the ip helper-address

<ip-address> command on the Voice VLAN interface of the router so that it will forward DHCP broadcasts from the

phones to the DHCP server.

Network Time Protocol Clock synchronization is important in VoIP systems for accurate Call Detail Records (used for billing), easier trou

bleshooting and debugging, and for good voice performance. Network Time Protocol (NTP) is used on all Cisco devices

to sync the system clock to a master clock. IP Phones get their time from the call agent (CM, CM Business Edition, CM

Express, or SBCS). The call agent(s) are configured to get their time from a master clock, usually a highly accurate

atomic or radio clock external to the network.

R o u t e r ( c o n f i g ) # c l o c k t imezone ps t -8

! S p e c i f i e s the l o c a l t imezone as PST (8 hours behind GMT)

R o u t e r ( c o n f i g ) # c l o c k summer-time zone r e c u r r i n g f i r s t Sunday a p r i l 02 :00 l a s t Sunday October 02:00

! A c t i v a t e s Summer Time change in the s p e c i f i e d date range

!

R o u t e r ( c o n f i g ) # n t p se rve r 1 0 . 1 . 2 . 3

! I d e n t i f i e s t h e NTP master c l o c k address



Cisco IP Phone Firmware and XML Configuration Files Cisco IP Phones need the following three separate files to function:

• The firmware file: This file is loaded into nonvolatile memory and is persistent across reboots. To make the

firmware files available to the phones, use the router command tftp-server flash:firmware-file-name. The command

load phone-type firmware-file is also required to associate the model of IP phone with the appropriate firmware file.

• SEPAAAABBBBCCCC.cnf.xml: This is the device-specific XML configuration file (AAAABBBBCCCC is the

MAC address of the phone), which specifies the IP address, port, firmware, locale, directory URL, and many other

pieces of information. This file is created when the IP Phone has been added to the configuration.

• XMLDefault.cnf.xml: This is the X M L configuration file that devices use if their specific SEP<MAC> file is not

available (typically if they have not registered before or if they have been factory reset).

These files are downloaded by the phone during its boot process.

Power over Ethernet Power over Ethernet (PoE) is a desirable option because it eliminates the cost and clutter of power bricks for the IP

Phones. There are two methods of PoE delivery:

• Cisco prestandard (inline power)

• 802.3af standard

Extra care should be taken to ensure the following:

• RJ-45 cabling is tested and meets the required standard.

• The IP Phone and the switch have a common PoE delivery method.

• The PoE switch has a suitable UPS backup to provide power continuance in the event of a power failure.

http://SEPAAAABBBBCCCC.cnf.xml

http://XMLDefault.cnf.xml


Alternatively, each IP Phone may be powered by its own cable and transformer, or a variety of power injectors are avail

able.

The Cisco prestandard PoE method works as follows:

1. The switch sends a special tone, called a Fast Link Pulse (FLP), out of the port. The FLP goes to the powered

device, in this case an IP phone.

2. When unpowered, the PoE device has a physical link between the pin on which the FLP arrives and a pin that goes

back to the switch. This creates a circuit, resulting in the FLP arriving back at the switch. Non-PoE devices will not

have this link; the switch will therefore never receive the FLP from a device that does not require PoE.

3. When the switch receives the returning FLP, it applies power to the line.

4. The link comes up within 5 seconds.

5. The powered device (IP phone) boots.

6. Using CDP, the IP Phone tells the switch exactly how much power it needs. (Power requirements vary from device

to device.)

The 802.3af PoE standard works slightly differently. The standard requires that all eight pins in the RJ-45 cable be

present and punched down. The following describes the 802.3af PoE negotiation steps:

1. The switch applies constant DC power to all ports that may require PoE.

2. An 802.3af-compliant device will apply 25 ohms resistance across the DC circuit.

3. The switch detects the resistance and applies low-power PoE to the link.

4. The powered device (the phone) boots.

5. The phone uses CDP to specify its power needs.


Qual i ty of Serv ice Quality of service (QoS) is possibly the single most important feature to deploy to ensure a successful VoIP system. This

section defines and describes why QoS is needed and explains how to configure and deploy a QoS solution using both the

Modular QoS Command Line (MQC) and AutoQoS.

QoS Definition QoS is defined as

The ability of the network to provide better or "special" service to a set of users and applications at the expense of other users and applications.

Voice and video traffic is very sensitive to delayed packets, lost packets, and variable delay (jitter). The effects of these

problems manifest as choppy audio, missing sounds, echo, or unacceptably long pauses in the conversation that cause

overlap, or one talker interrupting the other. QoS configurations provide bandwidth guarantees while minimizing delay

and jitter for priority traffic like VoIP. They do so not by creating additional bandwidth, but by controlling how the avail

able bandwidth is used by the different applications and protocols on the network. In effect, this often means that data

applications and protocols are restricted from accessing bandwidth when VoIP traffic needs it. This does not have much of

an impact on the data traffic, however, because it is generally not as delay or drop-sensitive as VoIP traffic.

The areas that QoS can address to improve and guarantee voice quality are the following:

• Bandwidth

• Delay (including delay variation or jitter)

• Packet loss


Bandwidth A VoIP call follows a single path from end to end. That path may include a variety of LAN and WAN links. The slowest

link represents the available bandwidth for the entire path—often referred to as a bottleneck because of the congestion the

slow link can cause.

If congestion is occurring, there are several ways to fix the problem:

• Increase the bandwidth: If bandwidth is unlimited, congestion cannot occur. Realistically, however, increasing

bandwidth is costly and is usually unnecessary if QoS is applied instead.

• Queuing: QoS employs advanced queuing strategies, which classify different traffic types and organize the classes

into queues that are emptied in order of priority. The queuing strategies commonly used in Cisco Unified

Communications include the following:

• Weighted fair queuing (WFQ): W F Q dynamically assigns bandwidth to traffic flows as they arrive at the

router interface. No configuration is necessary; the strategy is enabled by default on router links of Tl speed

and below. This strategy is not appropriate for VoIP because it does not provide a bandwidth guarantee for the

voice traffic, but instead allocates bandwidth fairly based on flow sizes (hence the name). VoIP needs a Priority

queue (PQ) that guarantees it the bandwidth it needs, at the expense of all other traffic.

• Class-based weighted fair queuing (CBWFQ): CBWFQ extends the W F Q algorithm to include user-defined

classes for traffic. Instead of the router dynamically interpreting traffic flows and building queues for them, the

admin classifies the traffic and assigns it to queues of configurable size and bandwidth allocation. There is still

no priority queue, however, so CBWFQ is not appropriate for VoIP.

• Low-latency queuing (LLQ): LLQ extends the CBWFQ system with the addition of a PQ. The PQ is typically

reserved for voice traffic, and if any packets show up in the PQ, all packets in the queue are immediately sent

while packets of other traffic types are held in their respective queues. This is the preferred queuing method for

VoIP networks.


• Compression: Several strategies are available to make the data that needs to be sent smaller so that it consumes less bandwidth:

• Payload compression: By compacting the contents of a packet, the total size is somewhat reduced. This

compression method does not affect the headers, which makes it appropriate for links that require the header to

be readable to route the packet correctly (Frame Relay and ATM as examples).

• Link compression: On point-to-point links where the header information is not needed to route the packet, link

compression may be used.

• Header compression: By specifying the use of compressed RTP (cRTP), the Layer 3 and 4 headers of a VoIP

packet are dramatically reduced, from 40 to as little as 2 bytes. TCP header compression is also available for

non-VoIP traffic using TCP transport.

Compression takes time and CPU resources, adding delay; this must be factored in to the decision of what strategies are

appropriate for a given link.

Delay Reducing end-to-end delay is a primary goal of QoS. Delay is calculated by adding the cumulative delay totals from

source to destination and will be expressed as one-way or round-trip. Delay is classified in the following ways:

• Fixed delay is predictable and constant. Sources of fixed delay include the following:

• Propagation delay: The amount of time it takes for the signal to transit the link. This is effectively the speed of

light as it moves through copper or optical media. Light travels just less than a foot in one-billionth of a second,

so long-distance links can impose significant delay that cannot be eliminated. L.A. to New York links routinely

see 40 ms one-way propagation delay.


• Serialization delay: This is the time it takes to load bits onto the media; this relates directly to the speed of the

link and cannot be altered unless that speed is changed.

• Variable delay includes processing and queuing delays; these will vary depending on the traffic load, the router

performance, and many other factors that are not easily predictable or constant.

Minimizing delay employs the same strategies as improving bandwidth:

• Increase link speed.

• Use Priority queuing (such as LLQ) for delay-sensitive traffic.

• Employ appropriate compression techniques.

Packet Loss Ideally, no packets of any type will be lost, but this is not realistic. We do need to minimize packet loss for VoIP traffic

because it has no mechanism to retransmit lost packets (unlike TCP, for example). Packets are lost for a variety of

reasons:

• Tail drop: When an output queue is full, no more arriving packets can be placed in the queue. Any packets that

arrive when the queue is full are dropped from the last position (tail) of the queue and cannot be recovered. This is

the most common source of packet loss.

• Input drop: If the input queue fills up, arriving packets are dropped and lost. This is rare, and it usually indicates an

overloaded router CPU.

• Overrun: Also the result of CPU congestion, overruns happen when the router cannot assign the packet to a free

buffer space.


• Ignore: There is no buffer space available.

• Frame errors: Problems in transmission created CRC errors, giant or runt frames. This is usually related to EMI or

failing interface hardware.

Minimizing loss can be achieved with QoS mechanisms like LLQ and compression or by increasing link speed. Some

additional and complementary strategies known as Link Efficiency mechanisms will help to prevent congestion:

• Traffic shaping: Delays packets and sends them at a configured maximum rate. For example, if an FTP server is

generating a 512 kbps stream, shaping could limit the output to 256 kbps, delaying the transmission of the excess

traffic. This will add significant delay and might cause packets to be dropped, so it is not desirable to shape VoIP

traffic, but shaping data traffic is an effective tool to complement voice QoS settings.

• Traffic policing: Drops packets in excess of a configured threshold. These packets may be retransmitted if the traffic

is using TCP, but because VoIP does not, policing should not be applied to VoIP traffic. Again, policing is an effec

tive complement to QoS configurations.

QoS Requirements for VoIP There are some accepted targets for delay, loss, and jitter for VoIP traffic. These are the targets that QoS and Link

Efficiency mechanisms help us reach:

• Delay should be less than 150 ms one way.

• Jitter (the variation in the delay between packets) should be less than 30 ms one way.

• Packet loss should be less than 1 percent.

• Each VoIP call requires between 17 kbps and 106 kbps of priority bandwidth, depending on the codec, compression,

and Layer 2 in use; it also requires another 150 bps per call for signaling traffic.

QoS Requirements for Data Traffic Although data requirements are not as strict as those of VoIP, it is appropriate to classify the enterprise data traffic into

four or five classes and assign each a certain amount of bandwidth in its particular queue. The Cisco QoS classification

tools include Network-Based Application Recognition (NBAR), which makes it simple to classify traffic that would

otherwise be difficult or impossible.

The classifications you create will compose the QoS policy for the organization. The policy will reflect the actual needs

of both voice and data traffic on the network and will be a living document that will adapt to changes in the organization

and the applications it uses for business. A QoS policy is developed using the following process:

1. Perform a network audit to determine the current state of traffic on the network. Determine whether congestion prob

lems already exist, and list all applications discovered in current use.

2. Perform a business audit to determine where the applications in use fit into the business model. Some apps will be

characterized as critical to the business, some as routine, and some as trivial or perhaps even undesirable. It is not

uncommon to discover that the business executive had no idea some apps were in use, and a decision needs to be

made about how to treat all applications discovered, legitimate or otherwise.

3. Determine the level of service required for each app. This will range from Priority for voice and video, through

Mission Critical, Urgent, Routine and Scavenger, and even Disallowed. The names are not important; the ones used

here serve only to identify the relative importance of the apps as they are classified. Every business audit will gener

ate a slightly different picture of what is vital to the business and what is to be disallowed.


• The requirements for video are similar; bandwidth consumption is calculated as [video codec output] + 20 percent.

For example: a typical 384 kbps video stream should be allocated 460 kbps of priority bandwidth.


4. Build the classification scheme to match the audit findings. Use the business audit and the executive's decisions to

create a classification scheme that lines up with the business needs.

5. Define the QoS settings for each traffic class. This may include a minimum and maximum bandwidth allocation, a

priority for each class, queuing strategies, and link efficiency methods, as appropriate.

AutoQoS QoS configuration is one of the more advanced skills in the IOS CLI environment. Although it is essentially simple in

architecture, the many commands needed are intimidating and time consuming. AutoQoS is a feature available on voice-

enabled IOS platforms to greatly simplify and automate QoS configurations. AutoQoS generates traffic classes and

service policies using predefined templates, making an in-depth understanding of the commands unnecessary. In any envi

ronment where there is a lack of skill or time, AutoQoS is a benefit. The autogenerated configuration adapts to changes

(such as the relocation of an IP Phone) and is manually customizable to meet specific requirements after the automated

config is completed. Auto QoS is available on all voice-enabled routers and switches with the correct IOS feature set.

Router AutoQoS is limited to the following interfaces:

• Serial PPP or HDLC links

• Frame Relay point-to-point links only

• ATM PVCs, both low- and high-speed

QoS Trust Boundary One of the important concepts in QoS is the trust boundary—the point at which the QoS marking of a packet or frame is

believed by the switch or router. If it is trusted, the packet is treated according to the QoS marking and corresponding

policy. If it is not trusted, it may be re-marked and treated differently.

Ideally, we want to place the trust boundary as close to the source (the devices generating the traffic) as possible. This

means that the trust boundary should actually be between the IP Phone and the attached PC, because generally we do not


trust the PC, but we do trust the phone. If there is no phone, the trust boundary is between the switch and the PC. The

switch must be able to recognize and configure the trust boundary; if it cannot, we must move the boundary up to the

gateway router.

AutoQoS can automatically detect and configure the trust boundary by sensing a connected Cisco IP Phone and applying

the necessary QoS commands.

Configuring AutoQoS The single command auto qos voip [trust] [fr-atm] enables AutoQoS at the interface. The keyword trust causes the

DSCP markings of packets to be trusted for classification purposes. If trust is not configured, traffic is classified using

NBAR, and the packets are DSCP marked as appropriate. The fr-atm keyword is used on Frame Relay and ATM point-

to-point links. The AutoQoS configurations are based on the configured bandwidth of the interface when AutoQoS is first

run; lowering the bandwidth after AutoQoS is run will not change the AutoQoS configurations, so AutoQoS must be

removed and reapplied if the bandwidth statement is changed.

On a switch interface, the keyword [ciscophone] enables the trusted boundary feature when the switch detects a Cisco

phone through its CDP messages. When a phone is detected, the QoS marking of packets is trusted; when no phone is

detected, the markings are not trusted.

Using the [trust] keyword on a switch interface causes the inbound QoS marking of packets to be trusted regardless of

whether a phone is detected.


Introducing Cisco Unif ied Communicat ions Manager Express Unified CM Express is a router-based call agent that scales up to 240 phones, depending on platform capacity. The

system extends the benefits of Unified Communications to small businesses.

Unified CM Express supports a wide range of TP Phone, system, and trunk features, as well as voice-mail integration with

Unity, Unity Express, and third-party systems using H.323 or analog DTMF signaling. For a complete feature list, refer to

the Unified Communications Manager Express 4.2 Data Sheets on cisco.com.

Unified CM Express runs on the ISR platforms, including the 2800 and 3800 series, and on the 3700 series Multiservice

Access Routers. The appropriate IOS IP Voice feature set, along with IP Phone licenses and firmware, and flash and

RAM appropriate for the install are required. The optional GUI files may be installed for simplified configuration and

administration but are not required.

Unified CM Express supports all current-generation IP Phones.

Defining Ephone and Ephone-DN An ephone is an Ethernet phone, and an ephone-dn is an Ethernet phone directory number. In CM Express, an ephone is a

logical configuration and settings for a physical phone, and the ephone-dn is a destination number that can be assigned to

multiple ephones.

An ephone-dn always has a primary directory number, and it may have a secondary one as well. When you create an

ephone-dn, you can specify it as single line (the default) or dual line. A single line can terminate one call; a dual line can

terminate two calls at the same time. This is necessary for call waiting, consultative transfer, and conferencing features

to work. When you create an ephone-dn, the router automatically creates POTS dial peers to match. The following

http://cisco.com


configuration creates a dual-line ephone-dn with a primary and secondary number. The number 20 in the configuration is

the tag, which is simply a unique identifier:

Rou te r ( con f i g )#ephone -dn 20 d u a l - l i n e

Route r (con f ig -ephone-dn)#number 5309 secondary 8675309

There is a maximum number of ephone-dns that a given platform will support; this is controlled by the hardware capacity

and by licensing. The max-dn <max-dn-value> command must be set to create ephone-dns; the default is zero. Be aware

that the router immediately reserves memory for the number of dns you specify, whether they are created or not; you

should configure only what you will actually need.

An ephone is the logical configuration of a physical phone. Each ephone is given a tag to uniquely identify it. The MAC

address of the phone ties it to the ephone configuration. CM Express will detect all phone models except the 7914

sidecar, which must be specified manually. Each different model of IP Phone has a different number of buttons, to which

various functions can be applied; the top button is always numbered " 1 , " with the others following in numerical order.

The button command allows you to specify which button does what. The following configuration creates a basic ephone

for a 7960 with a 7914 sidecar; the button 1:20 command assigns button 1 the dn (5309) assigned to ephone-dn 20 from

the previous example:

r o u t e r ( c o n f i g ) # ephone 20

r o u t e r ( c o n f i g - e p h o n e ) # mac-address AAAA.BBBB.CCCC

r o u t e r ( c o n f i g - e p h o n e ) # t ype 7960 addon 1 7914

r o u t e r ( c o n f i g - e p h o n e ) # b u t t o n 1:20

Types of ephone-dns Six types of ephone-dns are configurable in CM Express:

• Single line: This ephone-dn creates a single virtual port. Although you can specify a secondary number, the phone

can terminate only one call at a time, so it cannot support call waiting. It should be used when there is one phone


button for each PSTN line that comes into the system. It is useful for things like paging, intercom, call-park slots,

MoH feeds, and MWI.

Rou te r ( con f i g )#ephone -dn 1

Route r (con f ig -ephone-dn)#number 1001

• Dual line: The dual-line ephone-dn can support two call terminations at the same time and can have a primary and a

secondary number. It should be used when a single button supports call features like call waiting, transfer, and

conferencing. It should not be used for lines dedicated to intercom, paging, MoH feeds, MWI, or call park. It can be

used in combination with single-line ephone-dns on the same phone.



• Dual number: This ephone-dn has a primary and secondary number, making it possible to dial two separate

numbers to reach the phone. It can be either a single- or dual-line ephone-dn; it should be used when you want to

have two numbers for the same button without using more than one ephone-dn.

• Shared ephone-dn: The same ephone-dn and number appears on two separate phones as a shared line, meaning that

either phone can use the line, but once in use the other cannot then make calls on that line. The line will ring on all

phones that share the ephone-dn, but only one can pick up. If the call is placed on hold, any one of the other phones

sharing the line can pick it up.

• Multiple ephone-dns on one or more ephones: This configuration allows multiple calls to the same extension to be

handled simultaneously on a single phone; for example, using three dual-line ephone-dns with the same number will

terminate six calls on the phone. By using multiple ephone-dns on multiple phones, all the phones can answer the

same number. This is not a shared line because the phones will ring in succession, and a call on hold can be

answered only by the phone that placed it on hold. Controlling the hunting behavior (the order in which buttons or

phones ring) is done with the preference and huntstop commands, as explained in the "Hunting Configuration"

section that follows.


• Overlay ephone-dn: An overlay consists of two or more ephone-dns (up to 25) applied to the same button; all these

ephone-dns must be either single or dual line. (You can't mix the types.) The call coverage is similar to a shared-line

setup, except that a call to the number on one phone does not block the use of the same number on another phone.

You can overlay up to 10 lines on a single button and then configure the same overlay set on 10 phones, with the

result that all 10 phones could answer calls to the same number. The button command with the overlay separator

creates the overlay set. The overlay separator can be o, which designates an overlay set without call waiting, or c,

which designates call waiting. The command button lo30,31>32,33,34,35 c o n f j g u r e s ephone-dns 30, 31, 32, 33, 34,

and 35 on button 1 without call waiting.

Hunting Configuration Hunting allows a call to search for an available line to ring. This is commonly used in environments where call coverage

is needed to answer the same number, such as a call center or help desk. The preference command sets the order in

which the call will be tried on a list of ephone-dns; the huntstop command stops the hunting when it reaches that

ephone-dn; from this point, it is typical to send the call to voice mail.

The default is huntstop enabled. This can prevent calls from rolling over to the next ephone-dn, so the no huntstop

command must be used to allow the desired hunting behavior.

If dual-line ephone-dns are configured, the default behavior is for the call to hunt from the first line to the second. This

causes the same phone to ring twice for the same call.

The following configuration creates an ephone with two ephone-dns that both terminate calls to 1003. The huntstop configuration sends calls to the first channel of ephone-dn 3, then the second channel of ephone-dn 3, then the first

channel of ephone-dn 4, then the second channel of ephone-dn 4.



R o u t e r ( c o n f i g - e p h o n e - d n ) # p r e f e r e n c e 0

Rou te r ( con f i g -ephone -dn )#no hun ts top




R o u t e r ( c o n f i g - e p h o n e - d n ) # p r e f e r e n c e 1

R o u t e r ( c o n f i g - e p h o n e - d n ) # h u n t s t o p

Rou te r ( con f i g )#ephone 3

Rou te r (con f ig -ephone)#mac-address AAAA.BBBB.CCCC

R o u t e r ( c o n f i g - e p h o n e ) # b u t t o n 1:3 2 :4

This is not necessarily the behavior we want; it is more common to use the second channel of an ephone-dn for transfer,

call waiting, or conferencing. We can force the call to hunt from channel 1 of the first ephone-dn directly to channel 1 of

the second ephone-dn instead, using the hunststop channel command. The following configuration will send the call

from channel 1 of ephone-dn 5 (on button 2) to channel 1 of ephone-dn 6 (on button 3), then to channel 2 of ephone-dn 6

(also on button 3):


Router (conf ig -ephone-dn#number 1004

Rou te r ( con f i g -ephone -dn#p re fe rence 0

Rou te r ( con f i g -ephone -dn#hun ts top channel



R o u t e r ( c o n f i g - e p h o n e - d n # p r e f e r e n c e 1

Rou te r (con f i g -ephone-dn#no hun ts top channel


Router (con f ig -ephone#mac-address AAAA.BBBB.CCCC

R o u t e r ( c o n f i g - e p h o n e # b u t t o n 2 :5 3 :6

In a call-coverage scenario, we would want the call to hunt to an agent who is not already on the phone. Here we config

ure the call to hunt from channel 1 on the first phone to channel 1 on the second phone:











Router (con f ig -ephone#mac-address AAAA.BBBB.CCCC

R o u t e r ( c o n f i g - e p h o n e # b u t t o n 2 :5


Router (con f ig -ephone#mac-address DDDD.EEEE.FFFF

R o u t e r ( c o n f i g - e p h o n e # b u t t o n 2 :6


Configuring CM Express to Support Endpoints In this section we explore three methods of configuring endpoints on a CM Express system: configuring optional settings,

rebooting IP Phones, and troubleshooting and verifying the configuration.

Providing Firmware IP Phone firmware files ship with the CM Express software or can be downloaded from cisco.com. For the router to serve

the firmware to the phones, the tftp-server fiashifilename command is used. You must enter this command for every

firmware file needed. Some phones require more than one file to be loaded—for example, the 791 IG requires six separate

files.

Telephony Service Configuration Manual setup of the CM Express system is done using the CLI. From the global config, the command telephony-service enables config-telephony mode. This prompt is where your first steps of defining the max-ephones and max-ephone-dn settings (described earlier) would take place.

Phone Firmware Loads The firmware files that were copied into Flash and made available to the phones via TFTP must be associated with the

phones; this is done using the load model firmware-file command. Filenames are case sensitive, and the file extension

should not be included in the command. (Tip: Use the Cut-and-Paste function of your terminal client to prevent annoying

typos!) For Java-based phones, it is only necessary to load the TERMnn.x-y-x-w.loads or SCCPnn.x-y-x-w.loads

firmware filename (without the .loads extension), although the other files must be available via TFTP. The following is a

sample command set:

l oad 7960-7940 P00303020214

load 7920 cmterm_7920.4.0-01-08

load 7941 TERM41.7-0-3-0S

http://cisco.com


Defining Source IP and Port The CM Express software uses SCCP to communicate with the phones. (SIP signaling is also possible but is not covered

in this document.) The command ip source-address ip-address [port port] defines the IP address of the router that will

be used as the source for SCCP messages. The default SCCP TCP port is 2000 and does not normally need to be

changed, but the option is available if the situation should require it.

Autoregistration The autoregistration function is enabled by default; this allows a phone to be discovered and registered to an available

ephone slot (provided the ip source-address command is configured). The command no auto-reg-ephone prevents a

phone from registering unless its MAC address is explicitly configured already. CM Express records the MACs of all

phones that attempt to register but are blocked by autoregistration being disabled; use the show ephone

attempted-registrations command to see the list and the clear telephony-service ephone attempted-registrations

command to see and clear the list.

Create XML Config Files The create cnf-files command takes the configurations (including the firmware load, the source IP address, and port we

just defined) and builds an XML config file for each phone. This is a necessary step, and one that you may repeat from

time to time, for example, if you upgrade firmware or make other changes to the phone configuration.

DID Configurations It is common to have a range of DID numbers (fully qualified E.164 numbers) that allow outside callers to reach internal

extensions directly; usually, the DIDs have the four-digit internal extension as the last four digits. CM Express supports

this configuration with the dialplan-pattern command. This function expands extension numbers to full E.164 numbers

and converts incoming E.164 numbers to local extensions. The command is also needed to register the range of numbers

the command specifies with a gatekeeper; in fact, once configured, the range is automatically registered if a gatekeeper

is configured. You can disable this with the no-reg keyword. The full syntax is dialplan-pattern tag pattern

Automated Deployment of Endpoints In some cases, it is desirable to automate the deployment of phones. The telephony-service auto-assign command will

dynamically create ephones as physical phones are connected to the system, assigning an available ephone-dn to the

ephone. The ephone-dns must all be the same type (that is, single line or dual line). You must have a range of ephone-dns

configured, but it is no longer necessary to create each ephone and associate it manually.

The auto assign start-dn to stop-dn [type phone-type] [cfw number timeout seconds] command syntax specifies the

range of ephone-dns to use for a given phone model, the Call Forward Busy number to use (typically the voice-mail port),

and timeout values. You can enter multiple commands to specify ranges for your different phone types; if no phone type

is specified, any phone that registers will be assigned an ephone-dn from the specified range. The 7914 sidecar is not

supported by this command; phones with this add-on must have it manually added. The auto-assign cannot be used for

ephone-dns that serve paging, MoH, intercom or MWI functions. Nor can it be used for shared-line implementations.

Changes must be performed manually at the CLI, followed by resetting the affected phones. The following is a sample of

how the command can be used:

t e l e p h o n y - s e r v i c e

auto ass ign 11 to 20 t ype 7920

auto ass ign 21 to 30 t ype 7940

auto ass ign 31 to 40 type 7960

auto ass ign 41 to 50

ephone-dn 1 d u a l - l i n e

number 5301


extension-length length extension-pat tern pattern [no-reg]. The pattern uses the same wildcards as dial peers. A

sample configuration to set up a dial plan pattern for extensions 5300-5399 and expand them to the DID range of

867-555-5300-867-555-5399 would look like this:

t e l e p h o n y - s e r v i c e

d i a l p l a n - p a t t e r n 1 86755553. . e x t e n s i o n - l e n g t h 4 ex tens i on p a t t e r n 5 3 . .


The preceding output assigns ephone-dns from 11 to 20 to 7920s, 21 to 30 to 7940s, 31 to 40 to 7960s, and 41 to 50 to

any other type of phone (including those already specified, if there are no more ephone-dns in their range).

Location Customization CM Express supports phone display language, time display, and ring cadence localization. The user-locale language-

code command will change the language displayed on all 7940 and 7960 phones; the 7920 is not affected and must be

configured with its individual language capability local to the phone. The network-locale language-code command will

change the call progress tones and ring cadence (again with the exception of the 7920).

Following are language codes supported for User Locale:

• DE: Germany

• DK: Denmark

• ES: Spain

• FR: France

• IT: Italy

• NL: Netherlands

• NO: Norway

• PT: Portugal

• RU: Russian Federation

• SE: Sweden

• US: United States (default)

• JA:Japan


Following are language codes supported for Network Locale:

• AT: Austria

• CA: Canada

• CH: Switzerland

• DE: Germany

• DK: Denmark

• ES: Spain

• FR: France

• GB: United Kingdom

• IT: Italy

• JA: Japan

• NL: Netherlands

• NO: Norway

• PT: Portugal

• RU: Russian Federation

• SE: Sweden

• US: United States (default)

To change the time display format, use the time-format {12 I 24} command. To change the date format, use date-format

{mm-dd-yy I dd-mm-yy I yy-dd-mm I yy-mm-dd}.


Rebooting IP Phones There are two commands available to reboot IP Phones, each with a slightly different behavior. The reset command

causes a hard reboot of the phone and invokes DHCP and TFTP. Use reset when changing firmware, user/network

locales, or URLs. The reset command can be executed to reset a single phone at the config-ephone prompt, or at the

config-telephony prompt to reset one or more phones. The full syntax is reset {all [time-interval] I cancel I mac-address

Isequence-all}. The command options work as follows:

• all: Resets all phones.

• time-interval: Changes the interval between the router resetting the phones in sequence (default = 15sec).

• cancel: Stops the reset process.

• mac-address: Resets a specific phone.

• sequence-all: The router waits for one phone to reset and reregister before resetting the next phone to prevent the

phones from overloading the TFTP server. This can be time consuming; the router waits 4 minutes as a timeout

before resetting the next phone, whether or not the reregistration of the previous has finished.

The restart command causes a soft (warm) reboot and is useful for minor configuration changes, such as buttons, lines,

and speed-dial modifications. This command can also be executed either at the config-ephone prompt or at the config-

telephony prompt. The syntax is restart {all [time-interval] I mac-address}, with the command parameters the same as

the reset command.

Troubleshooting Endpoints Check the following when troubleshooting:

• Verify IP addressing: Use the Settings button on the phone to check the configuration of the IP phone. The TFTP

Server IP should be the CM Express router.


• Verify the files in flash memory: Verify that the correct firmware files are in the flash memory of the Cisco Unified

Communications Manager Express router using the show flash command.

• Debug the T F T P server: Use debug tftp events to ensure that the Cisco Unified Communications Manager

Express router is correctly providing the firmware and XML files.

• Verify the firmware installation of the phones: Use the debug ephone register command to verify which

firmware is being installed.

• Verify that the locale is correct: Use the show telephony-service tftp-bindings command to view the files that the

TFTP server is providing.

• Verify the phone setup: Use the show ephone command to view the status of the ephones and whether they are

registered correctly.

• Review the configuration: Use the show running-config command to verify the ephone-dn configuration.


Implement ing Basic Voice Features A business phone system is expected to provide the following features:

• Music on Hold

• Call Forward

• Call Transfer

• Call Park

• Intercom

• Paging

• Call Pickup

• Call Blocking

• Directory Services

The next sections describe the configuration of these basic business telephony features in CM Express.

Music on Hold No one likes to be on hold, but having something to listen to makes it a little better and can even relay useful information

to the listener. Configuring Music on Hold (MOH) in CM Express is simple. First copy a .wav file to Flash (avoiding

copyright issues by using royalty-free recordings). Next, issue the command moh wavefilename.wav in config-telephony

mode. By default, the router will multicast the stream to 239.23.4.10:2000; if you need to change this default multicast

address (typically you do not), issue the command multicast moh ip-address port port-number.


Call Forward The user can configure call-forwarding of all calls using the phone softkey. Using the CLI, you can configure different

call-forwarding options at the config-ephone-dn prompt:

• call-forward all directory-number: Forwards all calls to the specified directory number.

• call-forward busy directory-number: Forwards calls to the specified number if the user is on the phone.

• call-forward noan directory-number timeout seconds: Forwards calls to the specified directory number if the user

does not answer the phone before the specified timeout.

• call-forward maxlength length: Restricts the number of digits specified for the call-forwarding number; this

prevents call forwarding to an international long-distance number, for example.

Call Transfer Users can transfer calls with the Transfer softkey; the administrator can configure how this transfer happens using the

transfer-system {blind I full-blind I full-consult llocal-consult} config-telephony command. The command options are

as follows:

• Blind: Calls are transferred immediately using a Cisco-proprietary method.

• Full-blind: Calls are transferred immediately using the H.450.2 standard.

• Full-consult: Calls are transferred with consultation (meaning the user must speak to the target of the transfer before

the call is released); uses the H.450.2 standard.

• Local-consult: Uses a proprietary transfer method; not commonly used.


Call Park Call park allows a user to hold a call but retrieve it from another location by dialing the call park extension. A call-park

extension is a "floating" ephone-dn that is not assigned to any ephone. Multiple calls can be parked at a single extension

and are retrieved by dialling the extension; calls are picked up in the order in which they were parked. The syntax is rela

tively complex and specifies several options:

• park-slot [reserved-for extension-number] [timeout seconds limit count] [notify extension-number [only]] [recall]

[transfer extension-number][alternate extension-number][retry seconds limit count].

• reserved-for: (Optional) Indicates that this slot is a private park slot for the phone with the specified extension

number as its primary line. All lines on that phone can use this park slot.

• timeout seconds: (Optional) Sets the Call Park reminder timeout interval, in seconds. The range is from 0 to 65535.

When the interval expires, the Call Park reminder sends a 1-second ring and displays a message on the LCD panel of

the Cisco IP Phone that parked the call and to any extension that is specified with the notify keyword. By default,

the reminder ring is sent only to the phone that parked the call. If the timeout keyword is not used, no reminder ring

is sent to the extension that parked the call.

• limit count: (Optional) Sets a limit for the number of reminder timeouts and reminder rings for a parked call. For

example, a limit of 10 sends 10 reminder rings to the phone at intervals that are specified by the timeout keyword.

When a limit is set, a call parked at this slot is disconnected after the limit has been reached. The limit range is from

1 to 65535 reminders.

• notify extension-number: (Optional) Sends a reminder ring to the specified extension in addition to the reminder

ring that is sent to the phone that parked the call.

• only: (Optional) Sends a reminder ring only to the extension that is specified with the notify keyword and does not

send a reminder ring to the phone that parked the call. This option allows all reminder rings for parked calls to be

sent to the phone of a receptionist or an attendant, for example.

• recall: (Optional) Returns the call to the phone that parked it after the timeout limits expire.


• transfer: (Optional) Returns the call to the specified number after the timeout limits expire.

• alternate: (Optional) Returns the call to a specified second target number if the recall or transfer target phone is in

use on any of its extensions (ringing or in conversation).

• retry seconds: (Optional) Sets the delay before another attempt to recall or transfer a parked call, in seconds. The

range is from 0 to 65535. The number of attempts is set by the limit keyword.

The following example creates four call-park slots. Ephone-dn 10 and 11 can be used by any extension. A call parked in

these slots will stay parked for 100 seconds and will send a notification every 10 seconds to the extension that parked it.

If the 100-second limit elapses, the parked call is automatically transferred to 5309; if 5309 is busy or does not answer, it

goes to 5310. Ephone-dn 12 and 13 are reserved for 5301 and 5302, respectively. After a call has been parked for 100

seconds, it will be disconnected.

ephone-dn 10

number 7000

p a r k - s l o t t imeou t 10 l i m i t 10 t r a n s f e r 5309 a l t e r n a t e 5310

ephone-dn 11

number 7001

p a r k - s l o t t imeou t 10 l i m i t 10 t r a n s f e r 5309 a l t e r n a t e 5310

ephone-dn 12

number 7002

p a r k - s l o t t imeou t 10 l i m i t 10 r e s e r v e d - f o r 5301

ephone-dn 13

number 7003

p a r k - s l o t t imeou t 10 l i m i t 10 r e s e r v e d - f o r 5302


Intercom An intercom is a one-way audio speed-dial. Commonly used by an executive to an admin assistant, it allows the user to

press a phone button and be directly connected to another user. The destination phone answers the call in muted speaker-

phone mode so that privacy is maintained. Any user could dial the intercom if the extension is known. To make it impos

sible for anyone to dial the intercom (except those phones configured to do so), the extension number of the intercom can

include the A, B, C, or D character. These characters were at one time part of the touchtone dialpad, but because they are

no longer on the phone itself, users cannot dial them; it is still possible to configure them in an ephone-dn, however. The

following configuration shows a typical intercom configuration, using the B digit as part of the intercom extension

number:

ephone-dn 10

number 5301

name "Tommy TuTone"

ephone-dn 20

number 5309

name "Jenny"

ephone-dn 51

number B5555

name "Tommy TuTone"

i n te rcom B5556 l a b e l "Tommy TuTone"

ephone-dn 52

number B5556

name "Jenny"

i n te rcom B5555 l a b e l "Jenny"

ephone 6

bu t t on 1:10 2:51

ephone 7

bu t t on 1:20 2:52


Paging Audio paging builds a one-way audio path from the speaker to a single phone, a group of phones, or combined groups of

phones. A paging group is created by configuring a dummy ephone-dn with the paging command and associating that

ephone-dn with one or more ephones using the paging-dn command. When a user dials the paging extension, all config

ured phones answer the call in muted speakerphone mode. The default transport is unicast, which limits paging to a

maximum of 10 targets; multicast is also supported. The command syntax to create the ephone-dn is the following:

R o u t e r ( c o n f i g - e p h o n e - d n ) # paging [ i p multicast-address p o r t udp-port]

The following shows the syntax for associating an ephone to the paging ephone-dn. Note the unicast keyword, which will

override the multicast configuration if the phone is not reachable by multicast:

R o u t e r ( c o n f i g - e p h o n e ) # pag ing-dn paging-dn-tag [ u n i c a s t ]

The following example sets up a single paging group:

ephone-dn 25

number 2525

name Paging Sh ipp ing

paging i p 2 3 9 . 0 . 1 . 2 5 p o r t 2000

ephone-dn 18

number 1818

ephone-dn 15

number 1515

ephone 1

mac-address AAAA.BBBB.CCCC

bu t t on 1:18

pag ing-dn 25

ephone 2

mac-address BBBB.CCCC.DDDD

b u t t o n 1:15

pag ing-dn 25

Combining Paging Groups The config-ephone-dn command paging-grouppaging-dn-tag, paging-dn-tag,... is used to create a combined paging

group from multiple, previously defined paging dns. This is useful to create a paging group that reaches all phones for

emergency use, or simply to combine other groups for paging.

There are three variations of call pickup:

• Directed call pickup: Any extension can pick up a call that is on hold on another directory number, without belong

ing to a pickup group.

• Group pickup: A user can pick up a call for another group if the user knows the group extension. If only one

pickup group is defined, users need only press the Pickup softkey, whether or not they are a member of a pickup

group.

• Local group pickup: Users can pick up a ringing extension in their own group using the Pickup softkey plus the

star key (*).

An ephone-dn is assigned to a pickup group with the command pickup-group number. The numbers are arbitrary, but

the leading characters must be unique to each group; for example, the group numbers 81 and 817 will both be interpreted

If the ringing extension is in the user's group, pressing the Pickup softkey will redirect the call to the user's phone. If the

ringing extension is in another group, the user must press the GPickup softkey and enter the group number of the ringing

extension.

Call Pickup



Call Blocking Call blocking prevents unauthorized use of phones, typically to specific number patterns or times of day. You can define

up to 32 patterns of digits to block and apply a time schedule to restrict calls to whatever schedule suits your needs. Call

blocking applies to all IP Phones (except analog FXS phones), but phones can be exempted from call blocking individu

ally. An override function exists, configurable with a PIN for authorized users.

The schedule can be configured by day or by date using the following config-telephony commands:

a f t e r - h o u r s day day s t a r t - t i m e s t o p - t i m e

a f t e r - h o u r s da te month da te s t a r t - t i m e s t o p - t i m e

When the after-hours schedule is in place, use the block command to activate call blocking:

a f t e r - h o u r s b lock p a t t e r n t a g pattern [ 7 - 2 4 ]

The patterns use the same syntax as dial plan patterns. Using the 7-24 keyword blocks the configured pattern 24 hours a

day, 7 days a week, and disables the override PIN functionality.

The following configuration sets up a call blocking plan for all calls outside of normal business hours of 8:00 a.m. to 5:00

p.m., Monday through Sunday, and the holidays for New Year's day, the Fourth of July, and Christmas day:

R o u t e r ( c o n f i g ) # t e l e p h o n y - s e r v i c e

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day mon 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day tue 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day wed 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day t h u 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day f r i 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day sa t 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s day sun 17:00 08:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s da te j an 1 00:00 00:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s da te j u l 4 00:00 00:00


R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s date dec 25 00:00 00:00

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s b lock p a t t e r n 9011 !

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s b lock p a t t e r n 9011!#

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s b lock p a t t e r n 9 1 [ 2 - 9 ] . . [ 2 - 9 ]

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s b lock p a t t e r n 91900

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s b lock p a t t e r n 91976

R o u t e r ( c o n f i g - t e l e p h o n y ) # a f t e r - h o u r s b lock p a t t e r n 9 [ 2 - 9 ] . . [ 2 - 9 ]

Exempting a phone from after-hours blocking is easily configured with the after-hours exempt command at the config-

ephone prompt. Adding a PIN is equally simple at the same prompt with pin pin-number.

Directory Services Users can access the list of numbers and names by pressing the Directory key. Directory entries are drawn from the

ephone-dn configuration if it includes a name entry. CM Express supports 100 directory entries of up to 32 characters,

with the name being up to 24 characters. The directory entries can be listed as first-name-first or last-name-first;

whichever method is chosen, the name under the ephone-dn configuration should match to avoid confusion. The config-

telephony command to specify how names in the directory shall be displayed is the following:

d i r e c t o r y { f i r s t - n a m e - f i r s t | l a s t - n a m e - f i r s t }

It is possible to configure a directory entry that is not an IP Phone. To create such an entry, use the following config-

telephony command:

d i r e c t o r y e n t r y { [ e n t r y - t a g number name name] | c l e a r }


Maintaining a CM Express System An IP Phone system needs regular attention to watch for unusual events or unhealthy trends. Day-to-day operations and

maintenance tasks include the following:

• Updating files on the router

• Configuring syslog logging

• Billing procedures

• Managing Call Detail Records (CDRs)

The next sections discuss these topics and configurations.

Managing Router Files The CM Express router will need routine updates applied to improve reliability, add features, or enhance security.

Whether the files are upgrades to the Cisco IOS, the Communication Manager Express application or GUI, phone

firmware or MO H files, the command to load them into the router is the familiar copy tftp flash syntax. (The TFTP

server must be active and accessible over the network, of course.) FTP is also supported if file sizes greater than 32 MB

are to be moved; a suitable account and password must be configured for FTP transfers.

CM Express software is available as a bundled, single .zip file containing all the files needed to run CM Express, includ

ing the GUI. This single file can be extracted on the TFTP/FTP server and the files downloaded to the router Flash

memory.


SYSLOG and SNMP MIB Support CM Express supports type 6 Syslog messages for IP Phone registration. After a Syslog server is configured using the

logging ip_address command and is available on the network, these messages can be viewed along with other messages

generated by the router, using the Syslog viewer of your choice. The following are the Syslog messages provided for IP

Phone registration events:

• %IPPHONE-6-REG_ALARM

• %IPPHONE-6-REGISTER

• %IPPHONE-6-REGISTER_NEW

• %IPPHONE-6-UNREGISTER_ABNORMAL

• %IPPHONE-6-UNREGISTER_NORMAL

SNMP allows network system administrators to monitor changes and events by way of messages sent to a monitoring

application. CM Express support for SNMP MIBs specific to IP telephony activities and events includes the following

three MIBs:

• Cisco-DIAL-CONTROL-MIB (CDR and call history)

• Cisco-VOICE-CONTROL-MIB (extends to telephony and VoIP dial peers and call legs)

• Cisco-VOICE-IF-MIB

These MIBs, along with CDR data, allow visibility into detailed information about both summary and specific call infor

mation.


Billing Support Billing support is provided by way of CDR records and H.323 start/stop time AAA messages to the syslog server or

billing application. If the user enters an account code using the Acct softkey during call setup or when the call is

connected, the account code is recorded in the CDR and added to the Cisco-VOICE-DIAL-CONTROL-MIB. The account

code can then be accessed by a billing application to determine how long a user was on the phone with each customer,

and billed accordingly.

Call Detail Records Call Detail Records (CDR) are created by default and recorded in memory for later review and analysis using either the

CLI or GUI. CDRs can optionally be sent to the Syslog server.

Unity Express Hardware Capacities

Cisco Unity Express Module Max. Mailboxes Max. Sessions Internal Card Storage Device Hours of Storage

CUE—AIM 50 4 or 6 Yes Flash 14

CUE—NM 100 8 No HDD 300

CUE—NM-Enhanced 250 16 No HDD 300

Unity Express includes both a GUI and TUI interface, for initial mailbox setup as well as ongoing maintenance. The TUI

includes a tutorial to make it simple for users to set up their own mailboxes (or General Delivery boxes for group accessi

ble mailboxes), eliminating much of the administrative overhead.

Some of the features offered by Unity Express include the following:

• Alternate greetings—Allows a user to add a special greeting for an extended absence

• Message tagging (private or urgent)

Implement ing Cisco Unity Express Unity Express is a richly featured voice-mail and auto-attendant application that is coresident in the router in either a

Network Module format or Advanced Integration Module format. Having a local voice-mail application is ideal for

smaller organizations as a standalone solution or to provide local voice-mail access in a branch office of a larger organi

zation without having to send the traffic across the IP WAN if bandwidth utilization is an issue.

Cisco Unity Express supports up to 250 mailboxes (and 300 users), dependent on hardware platform. It can provide

voice-mail and integrated messaging, but not Unified Messaging. There is no provision for a TDM interface to a legacy

PBX voice-mail system (because the hardware is internal to the router), and there is no provision for redundancy. Unity

Express is actually an embedded Linux operating system, with an Ethernet interface to the router platform. (This interface

is not visible physically or in the router configuration.)

The following table summarizes the capacities of the three Unity Express hardware platforms.



• Reply, forward, or save messages

• Pause, fast-forward, or rewind messages during playback

• Envelope information

• 0-to-Operator with definable destination extension for Operator

• Message Waiting Indicator (MWI)

• Mailbox Full notification

• VPIM compatibility for message interchange with other Unity Express systems (or any other VPIM-compliant

system)

Unity Express Auto Attendant An Auto Attendant is essentially an interactive answering machine. It answers incoming calls, but it goes beyond that by

listening to the callers' responses to questions or options and offering more choices or playing specific greetings. If you

have ever heard "Press 1 for English; Appuyez sur le 2 pour Francais," you have heard an Auto Attendant. In addition,

Auto Attendant allows callers to search for the number of the person they are calling by first or last name, and Time-of-

Day and Day-of-Week call routing, so that different greetings are played when the business is closed. For many busi

nesses, Auto Attendant can eliminate the need for a receptionist—or at least free the person up to do other tasks.

An Auto Attendant is a logical mapping of greetings, options, and responses. Creating one requires careful mapping of

the decision and response tree. The Cisco Unity Express Editor is a tool that aids and speeds this process. Using the tool,

administrators can create multiple customized Auto Attendant flows. Using familiar Windows GUI-based actions, admin

istrators can drag-and-drop steps into the AA tree. Unity Express can run multiple AA scripts at the same time, providing

for very flexible and detailed responses to customer calls. If the Unity Express Editor GUI tool is not available and

changes need to be made, a TUI interface is also available. This can be very useful, for instance, if the administrator


wakes up to a foot of new snow and has to call in to the system from home to record an emergency greeting that explains

that the business is closed that day because of the snowfall. Users also have access to a TUI that allows them to change

their personal mailbox greetings and set or record their alternate greetings.

Unity Express GUI Much of the administration of UE can be managed from the administrative GUI. This includes normal operation such as

setting passwords and PINs for users, setting up mailboxes, creating users and groups, setting up backup and restore oper

ations, and restarting the system. To use the GUI, open the Unity Express URL at http://module_ip_address from a

supported browser. A command-line interface is also available for initializing the system and for times when the GUI is

not available. Certain tasks must be executed through the CLI. These include software installation, upgrade and licensing,

monitoring system resources (CPU, memory), and troubleshooting tasks such as viewing Syslog and trace files. Currently

only English language support is offered for both GUI and CLI, although other languages are supported for the TUI and

Auto Attendant. CUE provides the capability to bulk import users from Communications Manager Express at the

command line or by using the GUI.

Unity Express Software Files Unity Express comes preloaded with software from the factory; however, if you must reload the software or perform

upgrades, both a TFTP and an FTP server are required. The TFTP server must hold the following files:

• cue-installer.nm-aim.3.1.1: This is the installer file for version 3.1.1. (Other versions will have appropriate file

names.)

• A license file: Various license files can be loaded, each allowing a specific number of mailboxes. The filenames

include the number of licenses the file provides; for example, cue-vm-license_25mbx_cme_3.1.1.pkg provides 25

licenses.

http://module_ip_address


The FTP server must hold the following files:

• cue-vm.3.1.1.pkg: This is one of two system software files.

• cue-vm-full-k9.nm-aim.3.1.1.prtl: This is the second system software file.

• cue-vm-installer-k9.nm-aim.3.1.1.prtl: This is the application installation utility

• A language file: For example, cue-vm-en_US-lang-pack.nm-aim.3.1.1.prtl.

Router Configuration Prerequisites Unity Express can be coresident in the CM Express router, or it can be installed in a different router. In either case, some

basic configurations must be applied to the host router:

• Routing: Regardless of which routing protocol is in use, the CUE router must be able to reach all networks that

include hosts it must contact (voice-mail users, call agents, and so on).

• IP addressing: In addition to any interfaces that require IP addresses for network connectivity, the CUE module

itself requires addressing to enable the "hidden" Ethernet link across the backplane. The recommendation is to set it

up as follows:

i n t e r f a c e Loopback 0

i p address 192 .168 .66 .1 255 .255 .255 .0

! De f ines a so f twa re i n t e r f a c e

i

i n t e r f a c e S e r v i c e - E n g i n e l / 0

! Th is is t he CUE module p h y s i c a l i n t e r f a c e

I

ip unnumbered Loopback 0


! Con f i gu res the module to use the so f twa re i n t e r f a c e IP

i

se rv i ce -modu le i p address 192 .168 .66 .2 255 .255 .255 .0

! De f ines the IP of t he CUE o p e r a t i n g system

I

se rv i ce -modu le i p d e f a u l t - g a t e w a y 192 .168 .66 .1

ip rou te 192 .168 .66 .2 255.255.255.255 s e r v i c e - e n g i n e 1/0

! Con f igu res r o u t i n g f o r t h e CUE system to reach the r o u t e r and the r e s t o f t h e

! ne twork .

The Service-Engine and the Service-Module must be on the same subnet; they are the two hosts in a dedicated,

"hidden" Ethernet link that exists only on the backplane between the CUE hardware and the software running on it.

The router sees the CUE module as a separate host, even though it is physically internal to the router.

• Create a SIP dial peer pointing at the CUE service-module. CME uses SIP to communicate with the CUE system, so

a SIP dial peer with the following specific configurations must be created, even if there are no other SIP connections

in the CM system:

d i a l - p e e r vo i ce 7000 v o i p

! Creates the v o i p d i a l peer

d e s t i n a t i o n - p a t t e r n 7 7 . .

! De f ines the d i g i t p a t t e r n o f the mai lboxes

sess ion p r o t o c o l s i pv2

! Sets SIP as t h e p r o t o c o l used to communicate w i t h the CUE d i a l peer

sess ion t a r g e t i p v 4 : 1 9 2 . 1 6 8 . 6 6 . 2

! I d e n t i f i e s t h e IP address of the CUE s e r v i c e - e n g i n e

d t m f - r e l a y s i p - n o t i f y

! Forces DTMF d i g i t s to be sent o u t - o f - b a n d as SIP NOTIFY messages i n s t e a d of i n -band

codec g711ulaw

[ 104]


! Sets the codec to G .711 , which is t he o n l y codec CUE suppor t s

no vad

! I t is recommended to d i s a b l e VAD f o r the CUE system d i a l peer .

• Configure ephone-dns for MWI on and off functionality. There are some unique characteristics of these specialized

DNs: the digit patterns should be unique in the system, of course, but in addition a pattern of dots must follow the

digits, and that number of dots must equal the number of digits used in the local dial plan. This means that if you use

four digits for the local dial plan, there must be four dots; if a five-digit plan is used, there must be five, and so on.

(All extensions in the local dial plan must use the same number of digits.) The resulting configuration will look

something like this:

ephone-dn 75

number 4 4 7 5 . . . .

mwi on

ephone-dn 76

number 4 4 7 6 . . . .

mwi o f f

• Router HTTP access must be configured to support using the web-based GUI administration interface for both CUE

and CME. (The GUI for CME is not covered in this document.) The following commands will enable the HTTP

server, define the path to the HTTP files, and configure authentication:

• Router(config)# ip http server: Enables the web server (it is disabled by default)

• Router(config)#ip http path flash: Sets the path to the http files as the root of the Flash directory

• Router(config)#ip http authentication {aaalenablellocalltacacs}: Sets the authentication type used when

logging on to the web interface

It is possible—and typically recommended—to define a Unity Express web interface administrator that is separate from

the router administrator. Often the router admin and the CUE system admin are not the same person, and the CUE admin


may not have the skills to administer the router. To prevent the CUE admin from inadvertently causing unwanted changes

to the router config, create a separate CUE web interface admin using the following command at the config-telephony-service prompt:

web admin system name username {password string | sec re t {0 | 5} s t r i n g }

The secret keyword encrypts the password in the router configuration. If you want to enter a plain-text password that

should be converted to an MD5 hash, use the secret 0 string command; if the password is already MD5 hashed, use the

secret 5 string command. After the initial CUE web admin account is created, the admin can create additional accounts

for the customer administrator and users using the GUI. (These accounts can also be created using the CLI.)

Setting Up Unity Express After it is installed in the router, the CUE module starts automatically when the router is powered on. The module will

generally take longer than the router to fully boot up, and AIM modules in particular are slower to boot than NM

modules.

Command-line access to the CUE system is gained with the privileged exec command service-module service-engine mod/0 session. For remote access, connect to the router CLI using SSH and enter this command. The exit command

returns you to the router CLI.

The following are some other useful CUE commands:

• offline: Takes the CUE system offline. This command will warn you that all calls will be terminated if you confirm

with a "y."

• restore factory default: Self-explanatory; you are prompted with a message that all configuration info and data will

be irrevocably deleted. If you confirm "y," you have a factory-defaulted CUE system.

Post-Installation Configuration Tool A new (or factory-defaulted) CUE system will run the Post-Installation Configuration Tool the first time you log in.

IMPORTANT::

IMPORTANT:: Welcome to Cisco Systems Se rv i ce Engine

IMPORTANT:: post i n s t a l l a t i o n c o n f i g u r a t i o n t o o l .

IMPORTANT::

IMPORTANT:: Th i s is a one t ime process which w i l l gu ide

IMPORTANT:: you th rough i n i t i a l se tup o f your Se rv i ce Engine.

IMPORTANT:: Once r u n , t h i s process w i l l have c o n f i g u r e d

IMPORTANT:: t he system f o r your l o c a t i o n .

IMPORTANT::

IMPORTANT:: I f you do not w ish to c o n t i n u e , the system w i l l be h a l t e d

IMPORTANT:: so i t can be s a f e l y removed f rom the r o u t e r .

IMPORTANT::

Do you wish to s t a r t c o n f i g u r a t i o n now ( y , n ) ? y

Are you sure ( y , n ) ? y

The system will now ask you a series of questions to provide the basic information needed to allow it to interact with the

network and let the administrator log in:

Enter Hostname

(my-hostname, or e n t e r to use se -10 -90 -0 -10 )

Enter Domain Name

(mydomain.com, or e n t e r to use l o c a l d o m a i n ) :

Using l oca ldoma in as d e f a u l t

IMPORTANT:

IMPORTANT:


http://mydomain.com

IMPORTANT:: of IP addresses l i k e 1 .100.10.205 f o r se r ve rs used by CUE. In o rde r

IMPORTANT:: to c o n f i g u r e DNS you must know the IP address of at l e a s t one of your

IMPORTANT:: DNS S e r v e r s .

Would you l i k e to use DNS f o r CUE ( y , n ) ? n

WARNING: If DNS is not used CUE w i l l r e q u i r e t h e use

WARNING: of IP addresses.

Enter IP Address of t h e Pr imary NTP Server

( IP add ress ,o r en te r t o b y p a s s ) : 1 0 . 9 0 . 0 . 1

Found se rve r 1 0 . 9 0 . 0 . 1

Enter IP Address of t he Secondary NTP Server

( IP address , o r e n t e r to bypass) :

The next questions set the location and time zone for the CUE system:

Please i d e n t i f y a l o c a t i o n so t h a t t ime zone r u l e s can be se t c o r r e c t l y .

Please s e l e c t a c o n t i n e n t or ocean.


8) B o l i v i a 25) Guyana 42) Suriname

9) B r a z i l 26) H a i t i 43) T r i n i d a d & Tobago

10) Canada 27) Honduras 44) Turks & Caicos Is

11) Cayman I s l a n d s 28) Jamaica 45) Un i ted S ta tes

12) C h i l e 29) M a r t i n i q u e 46) Uruguay

13) Colombia 30) Mexico 47) Venezuela

14) Costa Rica 31 ) Mon tse r ra t 48) V i r g i n I s l a n d s (UK)

15) Cuba 32) Ne ther lands A n t i l l e s 49) V i r g i n I s l a n d s (US)

16) Dominica 33) Nicaragua

17) Dominican Repub l i c 34) Panama

#? 45

Please s e l e c t one of the f o l l o w i n g t ime zone r e g i o n s .

1) Eastern Time

2) Eastern Time - Mich igan - most l o c a t i o n s

3) Eastern Time - Kentucky - L o u i s v i l l e area

4) Eastern Standard Time - I nd i ana - most l o c a t i o n s

5) C e n t r a l Time

6) C e n t r a l Time - Mich igan - Wiscons in border

7) Mountain Time

8) Mountain Time - south Idaho & east Oregon

9) Mountain Time - Navajo

10) Mountain Standard Time - A r i zona

11) P a c i f i c Time

12) A laska Time

13) A laska Time - A laska panhandle

14) A laska Time - A laska panhandle neck

15) A laska Time - west A laska

16) A l e u t i a n I s l a n d s



17) Hawai i

#? 11

The f o l l o w i n g i n f o r m a t i o n has been g i v e n :

Un i ted S ta tes

P a c i f i c Time

The re fo re TZ= 'Amer ica /Los_Ange les ' w i l l be used.

Loca l t ime is now: Mon Apr 28 11 :01 :20 MST 2008.

U n i v e r s a l Time is now: Mon Apr 28 17 :01 :20 UTC 2008.

Is t he above i n f o r m a t i o n OK?

1) Yes

2) No

#? 1

C o n f i g u r i n g the sys tem. Please w a i t . . .

w a i t i n g 125 . . .

The next questions and answers create the administrator account and password:

IMPORTANT::

IMPORTANT:: A d m i n i s t r a t o r Account C r e a t i o n

IMPORTANT::

IMPORTANT:: Create an a d m i n i s t r a t o r accoun t . Wi th t h i s account ,

IMPORTANT:: you can l o g in to the Cisco U n i t y Express GUI and

IMPORTANT:: run the i n i t i a l i z a t i o n w i z a r d .

IMPORTANT::

Enter a d m i n i s t r a t o r user I D :

(user ID ) :Un i t yAdmin

Enter password f o r :

(password ) :C isco

Conf i rm password f o r b y r e e n t e r i n g i t :

(password) :Cisco

SYSTEM ONLINE

CUE>

At this point, you should be able to ping the IP address given to the CUE module from the PC you intend to use to

administer it. Open a supported web browser and go to http://cue_ip_address/. The first time you log in, a message

displays stating that only Administrator logins are allowed (until other users have been configured on the system). There

are several links to choose from; we are going to examine the Initialization Wizard.

CUE Initialization Wizard The Initialization Wizard allows you to quickly set up a brand-new (or factory-defaulted) CUE system.

F IGURE 1 2 Cue Initialization Wizard Login Screen


http://cue_ip_address/

At the opening screen (see Figure 12), the message in red clearly indicates that the system has not been configured and

that only Administrator logins are allowed. Log in with the credentials you supplied earlier.

F IGURE 1 3 Cue Initialization Wizard Entry Screen


F IGURE 1 4 Cue Initialization Wizard CM Login Screen

C i s c o C a l l M a n a g e r E x p r e s s

<6» > Powered bv-Cfscc -11 • - • 11 -C I S C O

Cisco Unity Express Initialization Wizard

Ca l lMan jg t r Express Login

Enter trie details of the CallManager Express that Cisco Unity Express will connect to The user name and password will be used to authenticate while retrieving information from the CallManagei Express

Hostname': '101 10.2

User Name *: jCisco

* indicates a mandatory field

:- | Next | : j Cancel | Help |

The CM Express Login page lets you provide the address and credentials the CUE unit will use to contact the CME

router. This is the IP address of the service engine.

F IGURE 15 Cue Initialization Wizard Import Users Screen


CUE will automatically import the users defined under the ephones in CME. You can then select whether users should have a mailbox, whether they are a voice-mail Administrator, and whether to set CFNA and CFB.

F IGURE 1 6 Cue Initialization Wizard System Defaults Screen

The next screen configures the system defaults for language, user passwords and PINs, mailbox and message max size,

and message retention window.


FIGURE 17 Cue Initialization Wizard Call Handling Screen

The Call Handling screen defines the DNs assigned for accessing voice mail, AA and the voice-mail operator, as well as

defining MWI operation.


FIGURE 1 8 Cue Initialization Wizard Commit Screen

Next, you are shown a review screen of the values you have entered so far, and you're given the option to commit the

changes or go back to modify them.

The final screen lists the committed information.


F IGURE 1 9 Cue Initialization Wizard Committed Information Screen

Auto Attendant The Auto Attendant (AA) is like the receptionist; a series of recorded messages and interactive prompts allows you to

create an answering system that gets callers either to an individual or to a voice mailbox so they can leave a message.

One advantage of having an AA is that it is then possible to free up the receptionist to do other useful tasks. An AA can

deal with multiple calls at the same time.


FIGURE 2 0 Auto Attendant List Screen

From the Voice Mail menu, select Auto Attendant. This shows a list of configured AAs. Clicking the name will lead you

to the configuration screens.

F IGURE 2 1 AA Language Settings

The first configuration is the language and script this AA will use.


118]

FIGURE 2 2 AA Scripts

The next screen allows you to choose the individual recordings that the script calls. It is also possible to record custom

AA script recordings.

FIGURE 2 3 AA Call Handling

The Call Handling screen lets you specify the extension the system will dial to reach this AA and how many concurrent

sessions the AA will support.



Introducing the Cisco Smart Business Communicat ions System The Cisco SBCS is a unified communications appliance aimed squarely at the small-business market. These all-in-one

devices support data, voice, video, AA, voice mail, security, and wireless for up to 50 users. They leverage UC500 Series

devices, including PoE switches, to provide the expansion capability to scale to the maximum endpoint capacity. Many

connectivity modules for WAN, Internet, and PSTN options are available, and a simple-to-use graphical interface configu

ration tool makes it cost effective for small businesses to take advantage of Cisco's Unified Communications products.

Hardware Components The core of the SBCS is the UC 500 Series for Small Business. This multiservice appliance incorporates routing, firewall,

VPN, IPS, PoE switchports, WAN and PSTN connectivity options, and wireless options. The SBCS incorporates CME

4.2 and CUE 3.1.1, with the features found on larger ISR hardware. The Catalyst 520 switch allows for expansion of the

system to support more endpoints than the UC500 core unit supports.

For more complex wireless deployments, the Cisco Mobility Express Solution with the Cisco 521 Wireless Express

Access Point and the Cisco 526 Wireless Express Mobility Controller provide scalable, manageable, and secure wireless

connectivity for both data and voice endpoints.

The SBCS supports a wide range of Cisco IP phones, including video and wireless capabilities. Specialized applications,

both from Cisco and third-party vendors, can integrate with the SBCS to further leverage the productivity gains offered

by unified communications.

The SBCS comes in two form factors: A desktop or wall-mount unit for installations of up to 16 users and a rack-mount

unit for 32-48-user deployments; the smaller units support ISDN BRI PSTN, FXO, and FXS connections, and the larger

units add support for Tl and El interfaces, both PRI and CAS.


Telephony Features The SBCS supports most of the features desired in a business phone system, including the following:

• PBX mode or keyswitch mode

• System features

• Language

• Date format

• System message

• System speed dials

• Network features

• Voice VLAN

• DHCP scope settings

• IP addressing

• SIP Trunk settings

• Dial Plan settings

• Extension length

• Outgoing call handling

• Incoming call handling

• Voice-mail features

• Voice-mail pilot numbers

• Auto Attendant


• Voice features

• MO H

• Paging

• Intercom

• Hunt Group

• Call Pickup

• Caller ID Blocking

• Call blocking

• Call Park

• Conferencing

• Users

• Name

• Association with a device

• Phone

• MAC address

• Extension number(s)

• Permissions

• Call Forward

Additional features are documented online.


Security Features The SBCS supports the Cisco IOS firewall, Easy VPN Server and Remote, NAT, and 802. lx authentication.

Wireless Features The smaller SBCS can be ordered with an integrated wireless AP, or external 521 Series wireless APs can be connected.

The larger SBCS models do not support internal APs. The standalone administrative capability of the Cisco Configuration

Assistant will support up to three connected APs. For support of up to 12 APs, the use of a Cisco 526 Wireless Express

Mobility Controller for every 6 APs is required. The SBCS systems provide full support for wireless security, including

WPA and WPA2, LEAP, PEAP, WEP, as well as voice VLANs with QoS.

Cisco Configuration Assistant The CCA is a powerful and simple GUI tool for administering the UC500 Series platforms. This tool is used to deploy,

configure, and maintain the SBCS devices, allowing control of the following:

• Switching

• Telephony

• Wireless

• Security

• Network services

• Internet connectivity

The GUI tool provides a network map view, showing the devices discovered in the system, as well as a front-panel view

of the SBCS system, showing ports and their status. The CCA even allows drag-and-drop upgrades to IOS software,

phone firmware, and language files.


I n t roduc ing the C i sco S m a r t Bus iness C o m m u n i c a t i o n s S y s t e m

FIGURE 2 4 Cisco Configuration Assistant Topology View

UC520

SEP000D29C0198C SEP0012D9FF3979 SEP001794627A1A SEP0017E06A3FCC

F IGURE 2 5 Cisco Configuration Assistant Front Panel View

Cisco Unified 5 0 0 Series

For those who miss the CLI, it is still possible to do all administrative tasks from the command line if you so desire.

Implementing Smart Business Communications System Voice Features The SBCS is remarkably simple to use; in fact, it ships with a default configuration that automatically assigns extensions

to phones as they are plugged in, enables the device to place and receive calls on the PSTN interface, and sets up default


configurations for the firewall, wireless (if applicable), NAT, VLANs, and telephony features. This is as close to a plug-

and-play phone system as it gets.

Install the Cisco Configuration Assistant on your administrative PC. (The installer is available as a free download from

Cisco.com.) When you run the software, it will ask for the IP address of the system to connect to; the default configura

tion gives the SBCS the IP of 192.168.10.1.

CCA will autodiscover any UC500 Series devices that are connected and generate a topology map. It is recommended,

however, that you use the Device Setup Wizard to perform the initial setup, because it integrates a number of setup proce

dures that are otherwise widely dispersed throughout the application. The following steps detail how to use the Device

Setup Wizard:

F IGURE 2 6 The CCA Device Setup Wizard— Step 1

1. Select a Device: With the CCA open, choose Setup, Device Setup Wizard. From the drop-down menu, choose the

device you want to configure. (Only devices in the UC500 Series will be available.) Click Next.

2. Prepare the Device: Verify that no other devices are connected; power them down or disconnect them if they are.

Click Next.


http://Cisco.com


3. Power Up Device: You are prompted to power up the device; if it is already powered up, click Next.

4. Connect Device to Your PC/Laptop: You must connect to one of the PoE ports with a straight-through Ethernet

cable. Wait until your PC has obtained an IP address; then click Next. The CCA will verify connectivity to the

device.

5. Verifying Connectivity: The CCA will contact the device and confirm connectivity to it. This may take a minute or

two.

6. Hostname and User Authentication: Enter the administrator username and password. Click Next.

7. Enter Date and Time Information: You have the choice of synchronizing the time to the PC's clock or setting it

manually. If you want to use NTP for the device's time synchronization, you can skip this step and configure NTP

later.

8. Enter IP Address and Other Device Setup Parameters: In this screen, you select the WAN interface and can then

choose to disable DHCP and set a static IP address.

9. Enter Other Device Setup Parameters: In this section, you select the Region, Phone Language, and Voicemail

language as appropriate to the device's location. These settings change the ring cadence on the phones as well as the

languages displayed and/or heard on the system.

10 . Summary: A brief summary of the configuration you have entered is displayed along with a brief caution that the

update may take up to 10 minutes.

When you launch the CCA application, the Connect window appears. Here you can enter a specific community, IP

address, or hostname of a device to connect to, modify options for connection port numbers, or create a new community

of devices.

A community is a group of SBCS devices (including 500 Series routers, 520 Series switches, wireless APs, and wireless

access controllers). The devices might not be in the same physical location or logical subnet. Communities make central

ized management of a related set of devices simpler; for example, if you have several customers, each of whom has an

SBCS system, you could create a community for each customer, making your administrative organization simpler.

CCA Menus After connection to the device or community, you have access to the menus in the left pane.

n $Q Selm

Device Setup Wizard..

F IGURE 2 8 The Setup Menu

The first menu is the Setup menu, under which is located the Device Setup Wizard detailed earlier.


F IGURE 2 7 The CCA Connect Window

FIGURE 2 9 The Configure Menu

Next is the Configure menu, which has options to configure Ports, Security, Telephony, Routing, Device Properties, and

Internet Connection. You can also save the system configuration from this menu.

Under Device Properties, the submenus include the following:

• IP Address: Allows you to view and change the IP addresses on a device and set DNS server IPs.

• Hostname: Allows you to change the hostname of a device.

• System Time: Allows you to view, set, and sync the time as well as configure NTP settings on one or more devices.

• HTTP Port: Allows you to change the HTTP default port the device uses.

• Users and Passwords: Here you can change the administrator password on a device or on all devices simultaneously.

• Device Access: Here you can set the allowed terminal protocols (Telnet, SSH, or both) that can be used to access a

device.


F IGURE 3 0 The Monitor Menu

SNMP: Here you configure SNMP settings, including location and contact, traps to send, community strings, and

so on.

F IGURE 3 1 The Maintenance Menu

The Monitor menu allows you to generate reports and change the view from Front Panel to Topology. The Health link

generates graphical charts showing the statistics for key performance counters. The Event Notification and System Event

Messages links allow you to view and acknowledge messages and resolve problems automatically using Cisco

Configuration Assistant (if possible).

The Maintenance menu gives you the ability to perform software upgrades (with drag-and-drop functionality), manage

the files stored on device Flash memory, manage your configuration archive, and restart or reset devices. The license

management option visible here may not be supported by the UC500 IOS in use, depending on the model.


Topology View

Under the Monitor menu, selecting Views, Topology generates a network map of the SBCS devices discovered or named

in the community. This view allows you to annotate devices with IP addresses, port IDs, a friendly name, or a MAC

address. Right-clicking or double-clicking a device allows you to view its properties or change the settings of devices; the

options vary depending on the device selected.

Front Panel View


Configure Menu: Telephony The Telephony menu includes the Voice configuration screen, which in turn includes several tabs. If there is an error or

missing information on any page, the tab will be highlighted in red. The following explains what is found and config

urable in each tab:

FIGURE 3 4 The Voice Device Tab

• Device: The Device tab allows you to modify the hardware configuration; however, this is seldom necessary because

it is autodiscovered by CCA. Here we can change the call agent from a PBX to a Key system depending on the

needs of the customer. (This decision is part of the planning process and will be largely made by the customer, with

advice from the designer.) This screen also lists the number of licenses (IP Phones) the unit supports.

Next under Views is the Front Panel View; this shows a graphical, interactive representation of the device. Interfaces are

configurable by right-clicking them; you can choose to configure all ports in a module or a single one. For example, you

can enable or disable the port and configure the duplex, speed, and PoE settings of an Ethernet port.


F IGURE 3 5 The Voice System Tab

System: The System tab lets you configure region, voice-mail, and phone language settings, clock format, and

system speed dials.


FIGURE 3 6

The Voice Network

Tab

• Network: This is where you configure the Voice VLAN and DHCP scope.


F IGURE 3 7 The Voice AA & Voicemail Tab

• AA & Voicemail: Here you set the extension numbers for the Auto Attendant and Voicemail, as well as their PSTN

access numbers.


F IGURE 3 8 The Voice SIP Trunk Tab

• SIP Trunk: SIP trunks are used to connect to other telephony devices or service providers. The SBCS provides

built-in support for AT&T and CBeyond Communications SIP trunking services, as well as generic SIP trunks for

other providers. On this page you identify the SIP Proxy and Registrar servers and the MWI server, define the digest

authentication username and password, and define domain information.

F IGURE 3 9 The Voice Features Tab


Voice Fea tu res : In this screen, you identify the M O H audio file, enable and configure Paging, Group Pickup,

Intercom, Hunt Groups, Call Park, and Conferencing. Additionally, you can configure the CallerlD Block code and

the Outgoing Call Block List.

FIGURE 4 0 The Dial Plan Tab

• Dial P lan : In the Dial Plan screen, you can adjust the number of digits per extension (the default is 3) and set the

numbering plan locale to North American or Other. Choosing North American preconfigures the area codes as three-

digit, the long-distance access code as 1, and the international code as Oi l—these are all standardized as part of the

North American Numbering Plan. Choosing Other allows you to customize the dial plan as needed for other number

ing plans worldwide. This page also allows you to configure the behavior for incoming calls; either send them to an

operator or have calls on a particular FXO port sent to a specific extension.


FIGURE 4 1 The Users Tab

• Users : Here you associate users with the phones discovered by CCA and add new phones as needed. You also have

access to the phone configuration screen by clicking the More button.


The More Screen

In this screen, you can change what the phone buttons do, configure user permission, define paging group, configure

intercom, and set timers and operations for busy and no-answer rules.

Implementing Additional Smart Business Communications System Features The SBCS includes support for many features beyond the telephone system; it is also a router, a firewall, an Ethernet

switch, a DHCP server, and optionally a wireless AP. This section will review the configuration of these elements.

Port Settings From the Configure menu, select Ports, Port Settings.


FIGURE 4 3 Port Settings Configuration

The Configuration Settings tab (shown in Figure 43) allows you to enable and disable ports, set duplex and speed, and

enable or disable PoE negotiation.


The Runtime Status tab shows what the port is actually doing. (In contrast to the setting of Auto in the Configuration tab,

here you see that ports have actually negotiated Full Duplex/100 Mbps and PoE.) At the top of the table you can see the

allocated PoE, expressed as Consumed and Remaining values. The display shows Unknown, Cisco, and IEEE under the

Device column; these relate to the different PoE delivery types (IEEE being the current standard, and Cisco being the

prestandard proprietary implementation. Unknown typically means the attached device does not need PoE).

Security

Under the Security menu, you will find submenus for NAT, VPN Server, Security Audit, and Firewall and DMZ.

NAT

Network Address Translation serves three purposes: First, it hides internal addresses from the outside network (typically

the Internet). Second, it can allow many internal addresses to access the Internet using a single, registered Internet IP.


F IGURE 4 5 The NAT Page

These first two capabilities are enabled by default on the SBCS. Third, it can provide selective access to internal IPO

addresses from the outside in a controlled manner; this is useful for reaching mail and FTP servers from the Internet, for

example.

The NAT page allows you to configure these specific server targets, as well as firewall service configuration.

VPN Server

The VPN Server page lists and allows you to create the user accounts that can access the system via VPN (to a maximum

of 10 concurrent sessions). You must define a preshared key, which is used in the authentication and encryption process.

Next, define the IP address range that will be assigned to remote clients connecting to the system. The option of enabling

Split Tunneling allows clients to use their own Internet connection for any network other than the ones listed; this is

commonly used if security is less of a concern.

Security Audit The Security Audit link allows you to inspect and report on the security configuration of a particular device. You are

presented with a list of security checks and an indication of whether the device has passed the check; from here, you can



select one or more checks and click OK to have the CCA fix the security problem automatically. Although it is conven

ient and simple, be aware that increasing the security settings of a device may block connectivity to some applications. If

this is the case, the change can also be undone in this interface, until the best course of action to both resolve the security

issue and allow the intended operation can be determined.

Firewall and DMZ The Firewall and DMZ page allows you to configure the basic security level (High, Medium, or Low) of the firewall to

apply a preconfigured set of typical restrictions, define which interfaces are trusted and untrusted, and also to define

which interface is the DMZ (Demilitarized Zone—a term that describes a screened network where certain servers and

resources are placed so that controlled access to them can be provided without risking the private network).

Routing Although the SBCS does not typically run dynamic routing protocols (being designed for smaller installations where such

power is not required or will be handled by other devices), you do have the ability to configure static routes to ensure the

device can reach remote subnets not directly connected.

DHCP Configuring a DHCP server allows the SNCS to allocate IP address, subnet mask, and default gateway values to hosts on

the LAN. The interface allows you to create a scope of addresses for each VLAN. (A typical system will have one VLAN

for the phones and at least one more for the data devices, such as PCs.) You can also configure static DHCP bindings (so

that you can predict what IP a given MAC address will be assigned) and which addresses or range of addresses will be

excluded from the DHCP scope. The SBCS DHCP server is suited to the task of a small network deployment and should

not be used for larger environments.

Smartports

F IGURE 4 6 Smartports

The Smartports feature allows for rapid configuration of common interface settings appropriate to different device types;

for example, selecting Switch or Router from the pull-down list associated with a port will activate the 802.1Q trunking

protocol; selecting IP Phone + Desktop will configure multiple-VLAN functionality and QoS settings. The interface also

allows you to view and set the Access (data) and Voice VLANs per port. You can also view the port configuration for the

entire device by clicking its image and then clicking Details.

Wireless If the SBCS is equipped with or connected to a wireless device, by selecting Configure, WLANs you can view and

change settings for the SSIDs for data and voice (for use with wireless IP Phones such as the 7920 and 7921). Selecting

an SSID allows you to view and configure the wireless settings for the SSID, including the following:

• Broadcas t in Beacon: Select whether to make the SSID visible to wireless devices.

• VLAN: Change the VLAN to which the SSID belongs.



• Securi ty Sett ings: Change from the default of no security to a setting that may include authentication, encryption,

or both, using WEP, LEAP, WPA, or WPA2, among others.

Internet Connection This screen allows you to view and change the settings for the WAN interface. You can enable or disable the interface,

specify the use of PPPoE if your Internet provider requires it, and choose the addressing method. DHCP can be used, or

if your service provider has allocated you a static IP, you can specify the IP, mask, and default gateway. If you have

selected PPPoE, you can choose IP Negotiated, which relies on the negotiation capabilities of PPPoE to determine an IP

address.

Save Configuration This simple screen allows you to save the configuration of one or all devices to NVRAM, making it the startup configura

tion at the next reboot of the device.

Maintaining a Smart Business Communications System Several tools are included in CCA to monitor and maintain the SBCS. The Monitor menu includes Reports, with links for

Inventory and VPN Status; Views, with links to Front Panel and Topology (discussed previously), Health; Event

Notification; and System Messages.

Health

F IGURE 4 7 The Health Display

Under the Monitor menu, select Health. This generates a graphical representation (shown in Figure 47) of the critical

general health statistics of the SBCS: Bandwidth Utilization, Packet Error Rate, PoE Utilization, Temperature, CPU

Utilization, and Memory Utilization. These stats are updated every minute. More information can be read in the Health

Details window, accessed by clicking the Details button.


FIGURE 4 8 Event Monitor

The Event Notification window allows you to acknowledge event notifications, tell CCA to take action where possible,

and turn off the Alert LED on SBCS switches.

System Messages This screen allows you to view system messages from all devices or any single device and apply filters for severity level,

if desired. These messages are the same that can be seen at the terminal monitor of the IOS CLI.


Event Notification CCA allows you to view Event logs for all devices in your network, with easy-to-read severity icons to quickly indicate

whether the event is serious or informational:

• Critical errors are marked as Level 0 and 1.

• Errors are marked as Level 2 or 3.

• Warnings are marked as Level 4.

• Informational events are marked as Level 5, 6, or 7.

The Filter button allows you to view only messages of the selected level(s). The icons for these messages are shown next:


Backup and Restore From the Maintenance menu, choose Configuration Archive. The window displays the Backup tab and the Restore tab. To

back up, select All Devices or just a single device. Add a descriptive note for future reference, make note of the backup

path, and click Back Up. The files are written to the C: \Documents and Settings\<user name>\ .configuration

assistants backups directory by default; by selecting the Preferences button, you can change the directory and choose

to save the configuration to the device before backing it up.

The Restore tab allows you to select your view of backed-up configurations:

• Show backed-up configurations of the selected device

• Show backed-up configurations of the selected device type

• Show all backed-up configurations

Choose a device to restore, select a backup file, note any descriptive comments, and click Restore.

Restart/Reset

Under the Maintenance menu, the next link is Restart/Reset. This allows you to reboot the chosen device and gives you

the option of resetting it to factory defaults if need be.

[ 147]

CCNA Voice Quick Reference Michael Valentine Copyright © 2008 Cisco Systems, Inc.

Published by:

Cisco Press 800 East 96th Street Indianapolis, Indiana 46240 USA

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