Ccna.Voice.Quick.Refence.Sheet

CCNA Voice Quick
Reference

Michael Valentine

ciscopress.com Your S h o r t Cut t o K n o w l e d g e

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 C T P 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.

A b o u t t h e T e c h n i c a l 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 Q u i c k Reference by Michael Valentine

Introduction

Introduction
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 administra-
tors 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 certifi-
cations 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 V o I P signaling protocols, Unified Communications Manager administration, traditional
telephony, gateway and gatekeeper configuration, Q o S , 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 V o I P back-
ground 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 certifica-
tion. 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 .

© 2 0 0 8 Cisco Systems Inc. All rights reserved. This publication is protected by copyright. Please see page 147 for more details.

CCNA Voice Quick Reference by Michael Valentine

Introduction

P u r p o s e of T h i s G u i d e
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.

W h o Should Read T h i s G u i d e
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 C C V P 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.

Introduction to Unified Communications
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.



Introduction
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.


FIGURE 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.


Smart Business Communications System

FIGURE 2
The Smart Business
Communications
System—Image ©
Cisco

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.


Unified Communication Manager Express

FIGURE 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 M C S 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 C M , 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 9 1 1 ,
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

Understanding Unified Communications
Applications
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.

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.
Max. T D M PBX
Product Users Messaging Capability Platform 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 Voice Mail + Integrated Messaging + MCS Yes Yes Yes
per server Unified Messaging

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


disk for greeting and message storage, and have greater capacity for storage than A I M 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 S M B , 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.


nified Communications Applications
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.

FIGURE 4
The Cisco
Telepresence 3000
System—Image ©
Cisco


Understanding Traditional Telephony
This section introduces traditional telephony systems, concepts, and applications.

The PSTN
FIGURE 5 Public Switched Telephone Network
A Representation of
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.


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

FIGURE 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



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



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.

0
FIGURE 7
PSTN Call Setup

Customer
Telephone


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.

Numbering Plans
NOTE A numbering plan is an organized distribution of telephone numbers administered by a regional or national authority. The
Codes do not always plan defines the rules that allocate numbers according to an established international telecommunications standard. For
need to be dialed; Local example, the North American Numbering Plan defines a country code of 1, followed by a three-digit area code, a three-
numbers must always be digit office code, and a four-digit local number. There are numerous other numbering plans for other countries or regions
dialed.
of the world.


The North American Numbering Plan
Let's look at the N A N P 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

NOTE 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
Several other ranges are through 9. You will never find an office or area code of OXX or 1XX; those numbers are either reserved for specialized
reserved for specialized purposes or would interfere with things like operator access numbers. Several ranges are also reserved for Easily
purposes. One commonly
Recognizable Codes (ERCs); these are numbers where the second and third numbers of the area code are the same. They
recognized one is 555-
are used for special services—for example, 800, 888, 877, and 866 are toll-free numbers. Another recognizable assign-
01XX, which is used in
film and TV, demonstra- ment is the " N i l " series: this includes 4 1 1 , 6 1 1 , and 911 numbers that are not used as area codes but for other special
tions, or education. No assignments, such as information or emergency services.
actual customer is
assigned these numbers,
so calling a number seen
in a movie will not pose a
E.164 Addressing
nuisance to anyone. When The E.164 addressing scheme is an international standard for telephone numbering plans, originally developed by the
Tommy Tutone recorded International Telecommunication Union. An E.164 number contains the following components:
"867-5309/Jenny," he
immediately annoyed CC—Country Code
thousands of phone
NDC—National Designation Code
customers worldwide.
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.


Introduction 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 F X O 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 F X O ports. These ports allow the gateway to
place and receive calls to/from the PSTN. FXO ports also include a codec.

FIGURE 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.


FIGURE 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.


Introduction 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.


FIGURE 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:

10011100

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.


NOTE Compression is not a required step, but it is often done to save bandwidth in VoIP environments. The two main types of
The determination of compression we are concerned with are the following:
voice quality is based on
the Mean Opinion Score • Adaptive Differential P C M (ADPCM): This method does not send entire code words, but instead sends a smaller
(MOS). This is a subjec- code that represents the difference between this word and the last one sent. This is not commonly used today,
tive measurement,
because it produces lower voice quality and compresses down only to about 16 kbps.
created by gathering the
opinion of live human • Conjugate Structure Algebraic Code Excited Linear Prediction (CS_ACELP): As the name suggests, this is
listeners. A sample more complex compression. Based on a dictionary or codebook of known sounds made by a standardized American
recording is played, and
male voice, the digital sample is analyzed and compared to the dictionary. The dictionary code that is the closest to
the listeners give it a
score out of 5, where 5 is the sample is sent. The codebook is constantly learning. The output of this compression is typically 8 kbps—with
best. The same sample is very little degradation of voice quality. This compression is widely used in VoIP.
played using different
compression or process-
ing methods and scored
again. Because MOS is Time Division Multiplexing (TDM)
so subjective, other
T D M is the primary technology used in traditional digital voice; it is also extensively used in data circuits. The basic
quality measurements
exist that are more
premise is to take pieces of multiple streams of digital data and interleave them on a single transmission medium.
empirical and more accu-
rate. For reference, stan- T1 Circuits
dard PCM encoding
(G.711) scores 4.1, and On a Tl circuit, there are up to 24 channels available for voice. 64k from conversation 1 is loaded into the first Tl
G.729 scores 3.92. 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 - 1 6 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 C A S ; 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 A B C D 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.


U n d e r s t a n d i n g 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.

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 D S P resources available to you as possible. The D S P 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. U D P provides
multiplexing and checksum capability; RTP provides payload identification, timestamps, and sequence numbering.

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 = 2 0 0 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 87.200 82,400 82,400 31,200 26,400 26,400
(@ 50 pps) (87.2 kbps) (82.4 kbps) (82.4 kbps) (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 12,000 bps 11,200
(68 kbps) (67.2 kbps) (12 kbps) (11.2 kbps)

Voice Activity Detection (VAD)
Phone conversations on average include about 3 5 % 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, D S P 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.


I n t r o d u c i n g 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
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 L A N


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 M G C P 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 M G C P gateway. M G C P is not as widely implemented as SIP or H.323. M G C P 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 T D M 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.


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 Provider
Network
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 T D M
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, F X S , 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


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 p o t s
Gateway(config-dialpeer)#destination-pattern 8675309
Gateway(config-dialpeer)#port 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
+ 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.


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 P r o v i d e r 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 voice 20 voip
Gateway(config-dialpeer)# destination-pattern 4....
Gateway(config-dialpeer)# session target ipv4:10.1.1.2

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).

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

NOTE 1. Look for the incoming called-number command in a dial peer that matches the called number or DNIS string of the
The default dial peer 0 inbound leg.
cannot be deleted or
modified. It does not 2. Look for the answer-address command in a dial peer that matches the calling number or ANI string of the inbound
negotiate services such call leg.
as VAD, DTMF Relay, or
3. Look for the destination-pattern command in a dial peer that matches the calling number or ANI string of the
TCL applications. The
dial peer 0 configuration inbound call leg.
for inbound VoIP calls 4. Look for the POTS dial peer port command that matches the voice port of the incoming call (POTS dial peers only).
contains the following:
• Any codec
5. If all of the above fail to match, match against Default Dial Peer 0 as a last resort.

• VAD enabled The default dial peer 0 config for inbound POTS calls includes the following:
• No RSVP Support
• no ivr application
• Fax-rate voice
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.


Consider the following configuration:

d i a l - p e e r voice 10 voip
destination-pattern .T
session t a r g e t ipv4:10.10.10.1
i
d i a l peer v o i c e 2 0 v o i p
destination-pattern 867[2-3]...
!
destination-pattern 8674...
session 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)

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 T D M links. Internet Telephony Service


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 A N I 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 voice 20 pots
destination-pattern 867....
port 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:
Dial-peer voice 1 voip
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 ipv4:10.1.1.1
!
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:

Destination pattern 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:

prefix 6045552
port 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.



forward-digits 7
port 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 . . . 5552...
port 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:
voice t r a n s l a t i o n - r u l e 1
rule 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.


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:
v o i c e t r a n s l a t i o n - p r o f i l e JENNY
translate called 1

3. Apply the profile to one or more dial peers, either inbound or outbound:
description translated t o Jenny
t r a n s l a t i o n - p r o f i l e outgoing JENNY
port 1/0/0

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

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.
A
[ 0-9] Do not match a single character specified in the list.

continues


Cisco Regular Expression Characters for Voice Translation Rules continued
* 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:

rule 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:
voice t r a n s l a t i o n ? r u l e 1
A
rule 1 / 4 0 . . . / /66660B0/

This example replaces any five-digit number that begins with " 4 0 " with the number "6666000".

Example 3:

voice t r a n s l a t i o n ? r u l e 1
A
/ (867)(....)/ /5552/


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 P L A R configuration that will call 867-5309 when the phone goes off-hook:

voice port 1/0/0
connection 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 v o i c e [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 y p e = v o i c e , information type = voice,
description = '',
tag = 6001, destination-pattern = "6001',
answer-address = '', preference=0,
CLID R e s t r i c t i o n = None
1
CLID Network Number = "
CLID Second Number s e n t
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 = target 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 = '', target trunk-group-label = ' ' ,
numbering Type = "unknown'
group = 6 0 0 1 , Admin s t a t e i s u p , O p e r a t i o n s t a t e i s up,
i n c o m i n g c a l l e d - number = connections/maximum = 0 / u n l i m i t e d ,
DTMF Relay = d i s a b l e d ,
<output omitted>


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

d i a l - p e e r hunt 0
PASS
TAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF THRU SESS TARGET
100 p o t s up up 0
101 v o i p up up 5550112 0 syst ipv4 10.10.1 1
102 v o i p up up 5550134 0 syst ipv4 10.10.1 1
99 v o i p up down 0 syst
33 p o t s 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 o n

*May 1 19:32:11.731: //-1/6372E2598012/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) a f t e r DP_MATCH_INCOMING_DNIS; Incoming D i a l - p e e r = 1 0 0
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
Result=Success(0) a f t e r DP_MATCH_INCOMING_DNIS; Incoming D i a l - p e e r = 1 0 0
*May 1 19:32:11.735: //-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


Match Rule=DP_MATCH_DEST; C a l l e d Number=3600
Result=Success(0) a f t e r DP_MATCH_DEST
*May 1 19:32:11.735: //-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

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 u p .

Applique type is Channelized T1
Cablelength is short 133
No a l a r m s d e t e c t e d .
Framing i s ESF, L i n e Code i s AMI, Clock Source i s 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 i n e 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,


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 P r o v i d e r N e t w o r k
0 L i n e E r r Sees, 0 Degraded Mins 0 E r r o r e d Sees, 0 B u r s t y E r r Sees,
0 S e v e r e l y 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 port 1/0/0

E&M S l o t i s 1 , Sub-unit is 0, Port is 0
Type of V o i c e P o r t is E&M
O p e r a t i o n 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 i s s e t t o 0 s
I n t e r d i g i t Time Out i s s e t t o 0 s


Analog Info Follows:
Region Tone i s s e t f o r n o r t h a m e r i c a

Voice card s p e c i f i c Info Follows:
S i g n a l Type is wink-start
O p e r a t i o n Type i s 2 - w i r e
E&M Type is 1
D i a l Type i s dtmf
In Seizure is inactive
Out S e i z u r e is 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 E x p . : 1001

VoiceEncapPeer1003
i n f o r m a t i o n type = v o i c e ,
t a g = 1003, destination-pattern = 1001',
answer-address = '', preference=0,
1
numbering Type = unknown'
group = 1003, Admin s t a t e i s u p , O p e r a t i o n s t a t e i s up,
incoming called-number = '', connections/maximum = 0 / u n l i m i t e d ,
DTMF Relay = d i s a b l e d ,
huntstop = enabled,
type = pots, prefix = ' ' ,


forward-digits default
1
session-target = ' , v o i c e - p o r t = '1/1'

debug voip ccapi inout
The debug voip ccapi inout c o m m a n d 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
c o m m a n d to understand how calls are being handled by the router. This c o m m a n d 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 o n

NOTE T h e following lines show information about the calling and called numbers. T h e network presentation indicator (NPI)
This debug generates a shows the type of transmission. The Incoming Dial-Peer field shows that the incoming dial peer has been matched.
very long output, which
is impractical to fully *Apr 18 2 0 : 4 2 : 1 9 . 3 4 7 : //-1/9C5A9CA88009/CCAPI/cc_api_call_setup_ind_common:

duplicate here. I suggest Interface=0x64F26F10, Call Info(
you familiarize yourself Calling Number=4085550111(TON=National, NPI=ISDN, S c r e e n i n g = U s e r , Passed,
with sample outputs from Presentation=Allowed),
the Cisco IOS Debug Called Number=83103(TON=Unknown, NPI=Unknown),
Command Guide or C a l l i n g Translated=FALSE, S u b s r i b e r Type S t r = R e g u l a r L i n e , FinalDestinationFlag=TRUE,
better yet from your own Incoming Dial-peer=1, P r o g r e s s I n d i c a t i o n = N U L L ( 0 ) , C a l l i n g I E Present=TRUE,
lab experimentation.
Source T r k g r p Route L a b e l = , T a r g e t T r k g r p Route L a b e l = , CLID T r a n s p a r e n t = F A L S E ) , C a l l
Id=-1
*Apr 18 2 0 : 4 2 : 1 9 . 3 4 7 : //-1/9C5A9CA88009/CCAPI/ccCheckClipClir:
I n : C a l l i n g Number=4085550111(TON=National, NPI=ISDN, S c r e e n i n g = U s e r , Passed,
Presentation=Allowed)
*Apr 18 2 0 : 4 2 : 1 9 . 3 4 7 : //-1/9C5A9CA88009/CCAPI/ccCheckClipClir:
Out: C a l l i n g Number=4085550111(TON=National, NPI=ISDN, Screening=User, Passed,
Presentation=Allowed)


Preparing the Infrastructure to Support Unified
Communications
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 V L A N
(VVLAN, also called an Auxiliary VLAN) is an additional V L A N for the exclusive use of VoIP and video traffic. The
benefits of using a V V L A N 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 V L A N 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 V V L A N ID using CDP messages, and the phone then sends frames from itself tagged
with the learned V V L A N ID and forwards frames from the attached PC untagged. These untagged frames will be tagged
with the access V L A N 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 V V L A N 100 and the PC in V L A N 50:

Switch(config)#interface FastEthernet 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 5 0
Switch(config-if)#switchport voice vlan 100
Switch(config-if)#spanning-tree portfast

DHCP
It is recommended that you use D H C P for IP Phone addressing. Create a separate subnet for the Voice V L A N and add the
Option 150 parameter to identify the T F T P server IP address. This can be done on an existing D H C P server, or a new one
can be added if necessary; Cisco routers have D H C P server capability. The following configuration is a typical example
of router-based D H C P to support IP Phones:

service dhcp
! e n a b l e s t h e DHCP s e r v i c e
I
i p dhcp e x c l u d e d - a d d r e s s 1 0 . 1 . 1 . 1 10.1.1.10

! 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 a s s i g n

ip dhcp p o o l name IP_PH0NES
! C r e a t e s a p o o l of a d d r e s s e s ( c a s e - s e n s i t i v e name) and e n t e r s DHCP c o n f i g u r a t i o n mode
I
network 1 0 . 1 . 1 . 0 2 5 5 . 2 5 5 . 2 5 5 . 0
! D e f i n e s t h e subnet o f a d d r e s s e s f o r t h e p o o l

default-router address 10.1.1.1
! D e f i n e s t h e d e f a u l t gateway


d n s - s e r v e r address 192.168.1.10 192.168.1.11
! i d e n t i f i e s t h e DNS s e r v e r IP a d d r e s s ( e s ) - up to 8 I P ' s
!
option 150 i p 192.168.1.2
! i d e n t i f i e s t h e TFTP s e r v e r IP

If you choose to use a D H C P server that resides on a different network, you will need to add the ip helper-address
<ip-address> command on the Voice V L A N interface of the router so that it will forward D H C P broadcasts from the
phones to the D H C P 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 timezone pst -8
! S p e c i f i e s t h e l o c a l t i m e z o n e as PST (8 hours b e h i n d 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 0 2 : 0 0 l a s t Sunday October 02:00
! A c t i v a t e s Summer Time change in t h e s p e c i f i e d d a t e range
!
Router(config)#ntp server 10.1.2.3
! 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 X M L configuration file ( A A A A B B B B C C C C 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.


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 F L P 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 F L P 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 C D P to specify its power needs.


Q u a l i t y of S e r v i c e
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 L A N 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): C B W F Q 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 C B W F Q is not appropriate for VoIP.

• Low-latency queuing (LLQ): L L Q extends the C B W F Q 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. T C P header compression is also available for
non-VoIP traffic using T C P 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 F T P 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.


• 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.

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.


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 H D L C 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 D S C P 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.


I n t r o d u c i n g Cisco U n i f i e d C o m m u n i c a t i o n s
M a n a g e r 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 D T M F 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
R A M 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


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:

Router(config)#ephone-dn 20 d u a l - l i n e
R o u t e r ( c o n f i g - e p h o n e - d n ) # n u m b e r 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 y p e 7960 addon 1 7914
router(config-ephone)# button 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,
M o H feeds, and MWI.
Router(config)#ephone-dn 1
Router(config-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, M o H 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 fjg u r e s ephone-dns 30, 3 1 , 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.

Router(config-ephone-dn)#preference 0
Router(config-ephone-dn)#no huntstop


Router(config-ephone-dn)#preference 1
Router(config-ephone-dn)#huntstop
Router(config)#ephone 3
Router(config-ephone)#mac-address AAAA.BBBB.CCCC
Router(config-ephone)#button 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(config-ephone-dn#number 1004
Router(config-ephone-dn#preference 0
Router(config-ephone-dn#huntstop channel
Router(config-ephone-dn#no huntstop channel
Router(config-ephone#mac-address AAAA.BBBB.CCCC
Router(config-ephone#button 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(config-ephone#mac-address AAAA.BBBB.CCCC
Router(config-ephone#button 2:5
Router(config-ephone#mac-address DDDD.EEEE.FFFF
Router(config-ephone#button 2:6


C o n f i g u r i n g C M Express t o S u p p o r t 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 T F T P 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 o a d 7960-7940 P00303020214
l o a d 7920 c m t e r m _ 7 9 2 0 . 4 . 0 - 0 1 - 0 8
l o a d 7941 TERM41.7-0-3-0S


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 X M L 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


extension-length length e x t e n s i o n - p a t t e r n 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:

telephony-service
dialplan-pattern 1 86755553.. e x t e n s i o n - l e n g t h 4 extension p a t t e r n 5 3 . .

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 M W I 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:

telephony-service
a u t o a s s i g n 11 to 20 t y p e 7920
a u t o a s s i g n 41 to 50

ephone-dn 1 d u a l - l i n e
number 5301


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 D H C P 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 T F T P 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 X M L 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.


I m p l e m e n t i n g Basic V o i c e F e a t u r e s
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 m o h 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 L C D 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 i m e o u t 1 0 l i m i t 1 0 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 i m e o u t 1 0 l i m i t 1 0 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 i m e o u t 1 0 l i m i t 1 0 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 i m e o u t 1 0 l i m i t 1 0 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 t e r c o m B5556 l a b e l "Tommy TuTone"
ephone-dn 52
number B5556
name "Jenny"
i n t e r c o m B5555 l a b e l "Jenny"
ephone 6
button 1:10 2:51
ephone 7
button 1:20 2 : 5 2


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:

Router(config-ephone-dn)# paging [ip multicast-address port 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:

Router(config-ephone)# paging-dn paging-dn-tag [unicast]

The following example sets up a single paging group:

ephone-dn 25
number 2525
name Paging S h i p p i n g
p a g i n g 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
button 1:18
p a g i n g - d n 25
ephone 2


mac-address BBBB.CCCC.DDDD
button 1:15
p a g i n g - d n 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.

Call Pickup
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 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 start-time stop-time
a f t e r - h o u r s d a t e month d a t e 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:

after-hours block pattern t a g pattern [7-24]

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:

Router(config)#telephony- service
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 1 7 : 0 0 0 8 : 0 0
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 u e 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 1 7 : 0 0 0 8 : 0 0
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 sat 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 date jan 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 date 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 d a t e dec 25 0 0 : 0 0 0 0 : 0 0
Router(config-telephony)#after-hours block pattern 9011!
Router(config-telephony)#after-hours block pattern 9011!#
Router(config-telephony)#after-hours block pattern 91[2-9]..[2-9]
Router(config-telephony)#after-hours block pattern 91900
Router(config-telephony)#after-hours block pattern 91976
Router(config-telephony)#after-hours block pattern 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:

directory {first-name-first | last-name-first}

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] | clear}


M a i n t a i n i n g a CM Express S y s t e m
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 M O 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 F T P 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

S N M P allows network system administrators to monitor changes and events by way of messages sent to a monitoring
application. CM Express support for S N M P 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 A A A 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 C D R 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.


I m p l e m e n t i n g Cisco U n i t y 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 T D M 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.

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)


• 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 U R L 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.


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 C U E 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 1 9 2 . 1 6 8 . 6 6 . 1 255.255.255.0
! Defines a software i n t e r f a c e
i

interface Service-Enginel/0
! T h i s is t h e CUE module p h y s i c a l i n t e r f a c e
I
ip unnumbered Loopback 0


! C o n f i g u r e s t h e module t o use t h e s o f t w a r e i n t e r f a c e I P
i
service-module ip address 192.168.66.2 255.255.255.0
! D e f i n e s t h e IP of t h e CUE o p e r a t i n g system
I
service-module ip default-gateway 192.168.66.1
ip route 192.168.66.2 255.255.255.255 service-engine 1/0
! C o n f i g u r e s r o u t i n g f o r t h e CUE system t o reach t h e r o u t e r and t h e r e s t o f t h e
! network.

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 C U E 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 C U E service-module. C M E uses SIP to communicate with the C U E 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 voice 7000 v o i p
! C r e a t e s t h e v o i p d i a l peer
destination-pattern 77..
! Defines the d i g i t p a t t e r n of the mailboxes
session protocol sipv2
! Sets SIP as t h e p r o t o c o l used to communicate w i t h t h e CUE d i a l peer
session target ipv4:192.168.66.2
! I d e n t i f i e s t h e IP a d d r e s s of t h e CUE s e r v i c e - e n g i n e
dtmf-relay sip-notify
! Forces DTMF d i g i t s to be s e n t o u t - o f - b a n d as SIP NOTIFY messages i n s t e a d of i n - b a n d
codec g711ulaw


[ 104]

! S e t s t h e codec to G . 7 1 1 , w h i c h is t h e o n l y codec CUE s u p p o r t s
no vad
! I t i s recommended t o d i s a b l e VAD f o r t h e CUE system d i a l p e e r .

• Configure ephone-dns for M W I 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 H T T P access must be configured to support using the web-based GUI administration interface for both CUE
and CME. (The GUI for C M E 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 C U E system admin are not the same person, and the C U E 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 C U E web interface admin using the following command at the config-telephony-
service prompt:

web admin system name username {password string | s e c r e t {0 | 5} string}

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 M D 5 hash, use the secret 0 string command; if the password is already M D 5 hashed, use the
secret 5 string command. After the initial C U E 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 C U E 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 C U E 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 C U E 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 C i s c o Systems S e r v i c e Engine
IMPORTANT:: p o s t installation configuration tool.
IMPORTANT::
IMPORTANT:: T h i s i s a one t i m e p r o c e s s w h i c h w i l l g u i d e
IMPORTANT:: you t h r o u g h i n i t i a l s e t u p o f y o u r S e r v i c e E n g i n e .
IMPORTANT:: Once r u n , t h i s p r o c e s s w i l l have c o n f i g u r e d
IMPORTANT:: t h e system f o r your l o c a t i o n .
IMPORTANT::
IMPORTANT:: I f you d o not w i s h t o c o n t i n u e , t h e system w i l l b e h a l t e d
IMPORTANT:: s o i t can b e s a f e l y removed f r o m t h e r o u t e r .
IMPORTANT::
Do you w i s h to s t a r t configuration now (y,n)?y
Are you s u r e (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:

E n t e r Hostname
(my-hostname, o r e n t e r t o use s e - 1 0 - 9 0 - 0 - 1 0 )
E n t e r Domain Name
(mydomain.com, o r e n t e r t o use l o c a l d o m a i n ) :
Using l o c a l d o m a i n a s d e f a u l t
IMPORTANT:
IMPORTANT:


IMPORTANT:: of IP a d d r e s s e s l i k e 1 . 1 0 0 . 1 0 . 2 0 5 f o r s e r v e r s used by CUE. In o r d e r
IMPORTANT:: to c o n f i g u r e DNS you must know t h e 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 a d d r e s s e s .

E n t e r IP Address of t h e P r i m a r y NTP S e r v e r
(IP address,or enter to bypass):10.90.0.1
Found server 10.90.0.1
E n t e r IP Address of t h e Secondary NTP S e r v e r
(IP address, or enter to bypass):

The next questions set the location and time zone for the C U E system:

Please i d e n t i f y a l o c a t i o n s o t h a t t i m e zone r u l e s can b e s e 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 o r o c e a n .


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) T u r k s & Caicos Is
11) Cayman Islands 28) Jamaica 45) U n i t e d States
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 R i c a 31 ) M o n t s e r r a t 48) V i r g i n Islands (UK)
15) Cuba 32) N e t h e r l a n d s Antilles 49) V i r g i n Islands (US)
16) Dominica 33) N i c a r a g u a
17) Dominican Republic 34) Panama
#? 45
Please s e l e c t one o f t h e f o l l o w i n g t i m e zone r e g i o n s .
1) E a s t e r n Time
2) E a s t e r n Time - M i c h i g a n - most l o c a t i o n s
3) E a s t e r n Time - Kentucky - L o u i s v i l l e a r e a
4) E a s t e r n S t a n d a r d Time - I n d i a n a - 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 - M i c h i g a n - Wisconsin border
7) M o u n t a i n Time
8) M o u n t a i n Time - s o u t h Idaho & e a s t Oregon
9) M o u n t a i n Time - Navajo
10) M o u n t a i n S t a n d a r d Time - Arizona
11) P a c i f i c Time
12) A l a s k a Time
13) A l a s k a Time - A l a s k a panhandle
14) A l a s k a Time - A l a s k a panhandle neck
15) A l a s k a Time - west A l a s k a
16) A l e u t i a n Islands


17) Hawaii
#? 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 :
United States
Pacific Time
Therefore TZ='America/Los_Angeles' w i l l b e used.
L o c a l t i m e is now: Mon Apr 28 1 1 : 0 1 : 2 0 MST 2008.
U n i v e r s a l Time is now: Mon Apr 28 1 7 : 0 1 : 2 0 UTC 2008.
Is t h e 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 system. Please w a i t . . .

waiting 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 account. With t h i s account,
IMPORTANT:: you can l o g in to t h e C i s c o U n i t y Express GUI and
IMPORTANT:: run t h e i n i t i a l i z a t i o n w i z a r d .
IMPORTANT::
E n t e r a d m i n i s t r a t o r user ID:
(user ID):UnityAdmin
E n t e r password f o r :
(password):Cisco


C o n f i r m 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 C U E 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) C U E system.

FIGURE 12
Cue Initialization
Wizard Login Screen


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.

FIGURE 13
Cue Initialization
Wizard Entry Screen


FIGURE 14 C i s c o CallManager Express -11 • - • 11 -
> Powered bv-Cfscc
Cue Initialization <6» CISCO

Wizard CM Login Cisco Unity Express Initialization Wizard

Screen
C a l l M a n j g 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 C M E
router. This is the IP address of the service engine.

FIGURE 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.

FIGURE 16
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 M W I operation.


FIGURE 18
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.


FIGURE 19
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 20
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.

FIGURE 21
AA Language
Settings

The first configuration is the language and script this AA will use.


118]

FIGURE 22
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 23
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.


I n t r o d u c i n g t h e C i s c o S m a r t Business
Communications 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 C M E
4.2 and C U E 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, F X O , 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 V L A N

• D H C P 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

• MOH

• Paging

• Intercom

• Hunt Group

• Call Pickup

• Caller ID Blocking

• Call blocking

• Call Park

• Conferencing

• Users

• Name

• Association with a device

• Phone

• M A C 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 C C A 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 C C A even allows drag-and-drop upgrades to IOS software,
phone firmware, and language files.



Introducing the Cisco Smart Business Communications System

FIGURE 24 UC520
Cisco Configuration
Assistant Topology
View

SEP000D29C0198C SEP0012D9FF3979 SEP001794627A1A SEP0017E06A3FCC

FIGURE 25
Cisco Configuration Cisco Unified 5 0 0 Series
Assistant Front Panel
View

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.

C C A will autodiscover any U C 5 0 0 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:

FIGURE 26
The CCA Device
Setup Wizard—
Step 1

1. Select a Device: With the C C A 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.


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 C C A 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 N T P 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 D H C P 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.

1 0 . 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.


FIGURE 27
The CCA Connect
Window

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.

C C A Menus
After connection to the device or community, you have access to the menus in the left pane.

FIGURE 28 n $Q Selm
The Setup Menu Device Setup Wizard..

The first menu is the Setup menu, under which is located the Device Setup Wizard detailed earlier.


FIGURE 29
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 N T P settings on one or more devices.

• H T T P 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.

S N M P : Here you configure S N M P settings, including location and contact, traps to send, community strings, and
so on.

FIGURE 3 0
The Monitor 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).

FIGURE 31
The Maintenance
Menu

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 M A C
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


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.

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 34
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.


FIGURE 35
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 36
The Voice Network
Tab

• N e t w o r k : This is where you configure the Voice V L A N and D H C P scope.


FIGURE 37
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.


FIGURE 38
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 M W I server, define the digest
authentication username and password, and define domain information.

FIGURE 39
The Voice Features
Tab


Voice F e a t u r e s : 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 40
The Dial Plan Tab

• Dial P l a n : 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 O i l — t h e s e 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 41
The Users Tab

• U s e r s : 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 D H C P 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 43
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 D M Z .

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.

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.

FIGURE 45
The NAT Page

The NAT page allows you to configure these specific server targets, as well as firewall service configuration.

VPN Server
The V P N 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 C C A 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 D M Z 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 D M Z (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 D H C P 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 D H C P 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 D H C P scope. The SBCS D H C P server is suited to the task of a small network deployment and should
not be used for larger environments.


Smartports

FIGURE 46
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, W L A N s 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:

• B r o a d c a s t in B e a c o n : Select whether to make the SSID visible to wireless devices.

• V L A N : Change the V L A N to which the SSID belongs.


• Security Settings: 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. D H C P 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 N V R A M , making it the startup configura-
tion at the next reboot of the device.

Maintaining a Smart Business Communications System
Several tools are included in C C A 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

FIGURE 47
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.


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:

FIGURE 48
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.


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: o u e t and Settings<user n m > .configuration
D c m ns a e
assistants b c u s directory by default; by selecting the Preferences button, you can change the directory and choose
akp
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]

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