VoIP Fundamentals
(Introducing Voice over IP Networks)
Voice over Internet Protocol (VoIP) allows a voice-enabled router to carry voice traffic, such as telephone calls and faxes,
over an Internet Protocol (IP) network. This topic introduces the fundamentals of VoIP, the various types of voice
gateways, and how to use gateways in different IP telephony environments.
Voice over IP is also known as VoIP. You might also hear VoIP referred to as IP Telephony. Both terms refer to sending
voice across an IP network. However, the primary distinction revolves around the endpoints in use. For example, in a VoIP
network, traditional analog or digital circuits connect into an IP network, typically through some sort of gateway. However,
an IP telephony environment contains endpoints that natively communicate using IP. Be aware that much of the literature
on the subject, including this topic, might use these terms interchangeably.
VoIP routes voice conversations over IP-based networks, including the Internet. VoIP has made it possible for businesses
to realize cost savings by utilizing their existing IP network to carry voice and data, especially where businesses have
underutilized network capacity that can carry VoIP at no additional cost. This section introduces VoIP, the required
components in VoIP networks, currently available VoIP signaling protocols, VoIP service issues, and media transmission
protocols.
Cisco Unified Communications Architecture
The Cisco Unified Communications System fully integrates communications by enabling data, voice, and video to be
transmitted over a single network infrastructure using standards-based IP. Leveraging the framework provided by Cisco
IP hardware and software products, the Cisco Unified Communications System has the capability to address current and
emerging communications needs in the enterprise environment. The Cisco Unified Communications family of products is
designed to optimize feature functionality, reduce configuration and maintenance requirements, and provide
interoperability with a variety of other applications. The Cisco Unified Communications System provides and maintains a
high level of availability, quality of service (QoS), and security for the network.
The Cisco Unified Communications System incorporates and integrates the following communications technologies:
■ IP telephony: IP telephony refers to technology that transmits voice communications over a network using IP standards.
Cisco Unified Communications System includes hardware and software products such as call processing agents, IP phones
(both wired and wireless), voice messaging systems, video devices, and other special applications.
■ Customer contact center: Cisco IP Contact Center products combine strategy with architecture to enable efficient and
effective customer communications across a global network. This allows organizations to draw from a broader range of
resources to service customers. These resources include access to a large pool of customer service agents and multiple
channels of communication as well as customer self-help tools.
■ Video telephony: The Cisco Unified Video Advantage products enable real-time video communications and collaboration
using the same IP network and call processing agent as Cisco Unified Communications. With Cisco Unified Video
Advantage, making a video call is just as easy as dialing a phone number.
■ Rich-media conferencing: Cisco Conference Connection and Cisco Unified MeetingPlace enhance the virtual meeting
environment with an integrated set of IP-based tools for voice, video, and web conferencing.
■ Third-party applications: Cisco works with other companies to provide a selection of third-party IP communications
applications and products. This helps businesses focus on critical needs such as messaging, customer care, and workforce
optimization.
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VoIP Overview
VoIP is the family of technologies that allows IP networks to be used for voice applications, such as telephony, voice instant
messaging, and teleconferencing. VoIP defines a way to carry voice calls over an IP network, including the digitization and
packetization of the voice streams. IP Telephony VoIP standards create a telephony system where higher-level features
such as advanced call routing, voice mail, and contact centers can be utilized.
VoIP services convert your voice into a digital signal that travels over an IP-based network. If you are calling a traditional
phone number, the signal is converted to a traditional telephone signal before it reaches its destination. VoIP allows you
to make a call directly from a computer, a VoIP phone, or a traditional analog phone connected to a special adapter. In
addition, wireless "hot spots" in locations such as airports, parks, and cafes that allow you to connect to the Internet might
enable you to use VoIP services.
Business Case for VoIP
The business advantages that drive the implementation of VoIP networks have changed over time. Starting with simple
media convergence, these advantages evolved to include call-switching intelligence and the total user experience.
Originally, ROI calculations centered on toll-bypass and converged-network savings. Although these savings are still
relevant today, advances in voice technologies allow organizations and service providers to differentiate their product
offerings by providing the following:
■ Cost savings: Traditional time-division multiplexing (TDM), which is used in the public switched telephone network
(PSTN) environment, dedicates 64 kbps of bandwidth per voice channel. This approach results in bandwidth being unused
when no voice traffic exists. VoIP shares bandwidth across multiple logical connections, which results in a more efficient
use of the bandwidth, thereby reducing bandwidth requirements. A substantial amount of equipment is needed to
combine 64-kbps channels into high-speed links for transport across a network. Packet telephony uses statistical analysis
to multiplex voice traffic alongside data traffic. This consolidation results in substantial savings on capital equipment and
operations costs.
■ Flexibility: The sophisticated functionality of IP networks allows organizations to be flexible in the types of applications
and services they provide to their customers and users. Service providers can easily segment customers. This helps them
to provide different applications, custom services, and rates depending on traffic volume needs and other customer-
specific factors.
■ Advanced features: Following are some examples of the advanced features provided by current VoIP applications:
■ Advanced call routing: When multiple paths exist to connect a call to its destination, some of these paths might be
preferred over others based on cost, distance, quality, partner handoffs, traffic load, or various other considerations.
Least-cost routing and time-of-day routing are two examples of advanced call routing that can be implemented to
determine the best possible route for each call.
■ Unified messaging: Unified messaging improves communications and productivity. It provides a single user interface for
messages that have been delivered over a variety of mediums. For example, users can read their e-mail, hear their voice
mail, and view fax messages by accessing a single inbox.
■ Integrated information systems: Organizations use VoIP to affect business process transformation. These processes
include centralized call control, geographically dispersed virtual contact centers, and access to resources and self-help
tools.
■ Long-distance toll bypass: Long-distance toll bypass is an attractive solution for organizations that place a significant
number of calls between sites that are charged traditional long-distance fees. In this case, it might be more cost-effective
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to use VoIP to place those calls across an IP network. If the IP WAN becomes congested, calls can overflow into the PSTN,
ensuring that no degradation occurs in voice quality.
■ Security: Mechanisms in an IP network allow an administrator to ensure that IP conversations are secure. Encryption of
sensitive signaling header fields and message bodies protect packets in case of unauthorized packet interception.
■ Customer relationships: The capability to provide customer support through multiple mediums, such as telephone, chat,
and e-mail, builds solid customer satisfaction and loyalty. A pervasive IP network allows organizations to provide contact
center agents with consolidated and up-to-date customer records along with related customer communication. Access to
this information allows quick problem solving, which builds strong customer relationships.
■ Telephony application services: XML services on Cisco IP Phones give users another way to perform or access business
applications. Some examples of XML-based services on Cisco IP Phones are user stock quotes, inventory checks, direct-
dial directory, announcements, and advertisements. Some Cisco IP Phones are equipped with a pixel-based display that
can display full graphics instead of just text in the window. The pixel-based display capabilities allow you to use
sophisticated graphical presentations for applications on Cisco IP Phones and make them available at any desktop,
counter, or location.
Components of a VoIP Network
Figure 1-1 depicts the basic components of a packet voice network.
Components of a VoIP Network
Figure 1-1 Components of a VoIP Network
The following is a description of these basic components:
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■ IP Phones: Cisco IP Phones provide IP endpoints for voice communication.
■ Gatekeeper: A gatekeeper provides Call Admission Control (CAC), bandwidth control and management, and address
translation.
■ Gateway: The gateway provides translation between VoIP and non-VoIP networks, such as the PSTN. Gateways also
provide physical access for local analog and digital voice devices, such as telephones, fax machines, key sets, and private
branch exchanges (PBX).
■ Multipoint Control Unit (MCU): An MCU provides real-time connectivity for participants in multiple locations to attend
the same videoconference or meeting.
■ Call agent: A call agent provides call control for IP phones, CAC, bandwidth control and management, and address
translation. Unlike a gatekeeper, which in a Cisco environment typically runs on a router, a call agent typically runs on a
server platform. Cisco Unified Communications Manager is an example of a call agent.
■ Application servers: Application servers provide services such as voice mail, unified messaging, and Cisco
Communications Manager Attendant Console.
■ Videoconference station: A videoconference station provides access for end-user participation in videoconferencing.
The videoconference station contains a video capture device for video input and a microphone for audio input. A user can
view video streams and hear audio that originates at a remote user station.
Other components, such as software voice applications, interactive voice response (IVR) systems, and soft phones, provide
additional services to meet the needs of an enterprise site.
VoIP Functions
In the traditional PSTN telephony network, all the elements required to complete a call are transparent to an end user.
Migration to VoIP requires an awareness of these required elements and a thorough understanding of the protocols and
components that provide the same functionality in an IP network.
Required VoIP functionality includes these functions:
■ Signaling: Signaling is the capability to generate and exchange control information that will be used to establish, monitor,
and release connections between two end-points. Voice signaling requires the capability to provide supervisory, address,
and alerting functionality between nodes. The PSTN network uses Signaling System 7 (SS7) to transport control messages.
SS7 uses out-of-band signaling, which, in this case, is the exchange of call control information in a separate dedicated
channel.
VoIP presents several options for signaling, including H.323, Session Initiation Protocol (SIP), H.248, Media Gateway
Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP). Some VoIP gateways are also capable of initiating SS7
signaling directly to the PSTN network. Signaling protocols are classified as either peer-to-peer or client/server protocols.
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Communications Manager Express System Administrator Guide
Cisco Unified Communications Manager Express (formerly known as Cisco Unified CallManager Express) is a call-
processing application in Cisco IOS software that enables Cisco routers to deliver key-system or hybrid PBX functionality
for enterprise branch offices or small businesses.
Cisco Unified CME Overview
Cisco Unified CME is a feature-rich entry-level IP telephony solution that is integrated directly into Cisco IOS software.
Cisco Unified CME allows small business customers and autonomous small enterprise branch offices to deploy voice, data,
and IP telephony on a single platform for small offices, thereby streamlining operations and lowering network costs.
Cisco Unified CME is ideal for customers who have data connectivity requirements and also have a need for a telephony
solution in the same office. Whether offered through a service provider’s managed services offering or purchased directly
by a corporation, Cisco Unified CME offers most of the core telephony features required in the small office, and also many
advanced features not available with traditional telephony solutions. The ability to deliver IP telephony and data routing
by using a single converged solution allows customers to optimize their operations and maintenance costs, resulting in a
very cost-effective solution that meets office needs.
A Cisco Unified CME system is extremely flexible because it is modular. A Cisco Unified CME system consists of a router
that serves as a gateway and one or more VLANs that connect IP phones and phone devices to the router.
Figure 2-1 shows a typical deployment of Cisco Unified CME with several phones and devices connected to it. The Cisco
Unified CME router is connected to the public switched telephone network (PSTN). The router can also connect to a
gatekeeper and a RADIUS billing server in the same network.
Figure 2-1 Cisco Unified CME for the Small- and Medium-Size Office
Figure 2-2 shows a branch office with several Cisco Unified IP phones connected to a Cisco IAD2430 series router with
Cisco Unified CME. The Cisco IAD2430 router is connected to a multiservice router at a service provider office, which
provides connection to the WAN and PSTN.
Figure 2-2 Cisco Unified CME for Service Providers
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A Cisco Unified CME system uses the following basic building blocks:
Ephone or voice register pool—A software concept that usually represents a physical telephone, although it is also used
to represent a port that connects to a voice-mail system, and provides the ability to configure a physical phone using Cisco
IOS software. Each phone can have multiple extensions associated with it and a single extension can be assigned to
multiple phones. Maximum number of ephones and voice register pools supported in a Cisco Unified CME system is equal
to the maximum number of physical phones that can be connected to the system.
Directory number—A software concept that represents the line that connects a voice channel to a phone. A directory
number represents a virtual voice port in the Cisco Unified CME system, so the maximum number of directory numbers
supported in Cisco Unified CME is the maximum number of simultaneous call connections that can occur. This concept is
different from the maximum number of physical lines in a traditional telephony system.
Licenses
You must purchase a base Cisco Unified CME feature license and phone user licenses that entitle you to use Cisco Unified
CME. In Cisco Unified CME Release 11, you should purchase:
Cisco Unified CME Permanent License
When you purchase a Cisco Unified CME permanent license, the permanent license is installed on the device when the
product is shipped to you. A permanent license never expires and you will gain access to that particular feature set for the
lifetime of the device across all IOS release. If you purchase a permanent license for Cisco Unified CME, you do not have
to go through the Evaluation Right to Use and Right To Use (RTU) licensing processes for using the features. If you want
to purchase a CME-SRST license for your existing device, you have to go through the RTU licensing process for using the
features. There is no change in the existing process for purchasing the license.
The Cisco Unified CME permanent license is available in the form of an XML cme-locked3 file. You should get the XML file
and load it in the flash memory of the device. To install the permanent license from the command prompt, use the license
install flash0:cme-locked3 command. The cme-locked3 is the xml file of the license.
Collaboration Professional Suite License
Collaboration Professional is a new suite of licenses. The Collaboration Professional Suite can be purchased either as a
permanent license or an RTU license. Collaboration Professional Suite Permanent License —When you purchase the
Collaboration Professional Suite license, by default, the Cisco Unified CME licenses are delivered as part of the
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Collaboration Professional Suite. You do not have to separately install and activate the Cisco Unified CME license. The
Collaboration Professional Suite permanent license is available in the form of an XML file. You should get the XML file and
load it in the flash memory of the device. To install the permanent license from the command prompt, use the license
install flash:lic_name command.
Collaboration Professional Suite RTU License—When you purchase the Collaboration Professional Suite RTU license, you
do not have to go through the Evaluation Right to Use process. However, you have to go through the RTU licensing process
for using the Cisco Unified CME features. To install the Collaboration Professional Suite RTU license from the command
prompt, use the license install flash0:colla_pro command. To activate the license, use the license boot module c2951
technology-package collabProSuitek9 command.
PBX or Keyswitch Model
When setting up a Cisco Unified CME system, you need to decide if call handling should be similar to that of a PBX, similar
to that of a keyswitch, or a hybrid of both. Cisco Unified CME provides significant flexibility in this area, but you must have
a clear understanding of the model that you choose.
PBX Model
The simplest model is the PBX model, in which most of the IP phones in your system have a single unique extension
number. Incoming PSTN calls are routed to a receptionist at an attendant console or to an automated attendant. Phone
users may be in separate offices or be geographically separated and therefore often use the telephone to contact each
other.
For this model, we recommend that you configure directory numbers as dual-lines so that each button that appears on an
IP phone can handle two concurrent calls. The phone user toggles between calls using the blue navigation button on the
phone. Dual-line directory numbers enable your configuration to support call waiting, call transfer with consultation, and
three-party conferencing (G.711 only).
Figure 2-3 shows a PSTN call that is received at the Cisco Unified CME router, which sends it to the designated receptionist
or automated attendant (1), which then routes it to the requested extension (2).
Figure 2-3 Incoming Call Using PBX Model
Keyswitch Model
In a keyswitch system, you can set up most of your phones to have a nearly identical configuration, in which each phone
is able to answer any incoming PSTN call on any line. Phone users are generally close to each other and seldom need to
use the telephone to contact each other.
For example, a 3x3 keyswitch system has three PSTN lines shared across three telephones, such that all three PSTN lines
appear on each of the three telephones. This permits an incoming call on any PSTN line to be directly answered by any
telephone—without the aid of a receptionist, an auto-attendant service, or the use of (expensive) DID lines. Also, the lines
act as shared lines—a call can be put on hold on one phone and resumed on another phone without invoking call transfer.
In the keyswitch model, the same directory numbers are assigned to all IP phones. When an incoming call arrives, it rings
all available IP phones. When multiple calls are present within the system at the same time, each individual call (ringing
or waiting on hold) is visible and can be directly selected by pressing the corresponding line button on an IP phone. In this
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model, calls can be moved between phones simply by putting the call on hold at one phone and selecting the call using
the line button on another phone. In a keyswitch model, the dual-line option is rarely appropriate because the PSTN lines
to which the directory numbers correspond do not themselves support dual-line configuration. Using the dual-line option
also makes configuration of call-coverage (hunting) behaviors more complex.
You configure the keyswitch model by creating a set of directory numbers that correspond one-to-one with your PSTN
lines. Then you configure your PSTN ports to route incoming calls to those ephone-dns. The maximum number of PSTN
lines that you can assign in this model can be limited by the number of available buttons on your IP phones. If so, the
overlay option may be useful for extending the number of lines that can be accessed by a phone.
Figure 2-4 shows an incoming call from the PSTN (1), which is routed to extension 1001 on all three phones (2).
Figure 2-4 Incoming PSTN Call Using Keyswitch Model
Figure 2-5 Incoming PSTN Call Using Hybrid PBX-Keyswitch Model
Call Details Records
The accounting process collects accounting data for each call leg created on the Cisco voice gateway. You can use this
information for post-processing activities such as generating billing records and network analysis. Voice gateways capture
accounting data in the form of call detail records (CDRs) containing attributes defined by Cisco. The gateway can send
CDRs to a RADIUS server, syslog server, or to a file in.csv format for storing to flash or an FTP server.
Cisco Unified CME
Cisco Unified CME 4.3 and later versions support the Cisco 1700 Model
The Cisco 1700 Router offers secure data, voice, and video communications with seamless mobility across wireless
networks independent of location or movement. This access router has a high-performance, compact, rugged design
optimized for use in vehicles in the defense, public safety, Homeland Security, and transportation markets.
Cisco Unified CME on the Cisco 1700 Series can be deployed in sites requiring on demand network connectivity and voice
and data communications that typically do not have PSTN connectivity.
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