Informational Signaling

DTMF tones are used not just for address signaling but also for informational signaling. Specifically, call-progress indicators in the form of tone combinations are used to notify subscribers of call status. Each combination of tones represents a different event in the call process, as follows:

• Dial tone Indicates that the telephone company is ready to receive digits from the user telephone. Cisco routers provide dial tone as a method of showing that the hardware is installed. In a PBX or key telephone system, the dial tone indicates the system is ready to receive digits.

• Busy Indicates that a call cannot be completed because the telephone at the remote end is already in use.

• Ringback (CO or PBX) Indicates that the telephone switch is attempting to complete a call on behalf of a subscriber.

• Congestion Indicates that congestion in the long-distance telephone network is preventing a telephone call from being processed. The congestion tone is sometimes known as the all-circuits-busy tone.

• Reorder Indicates that all of the local telephone circuits are busy, thus preventing a telephone call from being processed. The reorder tone is known to the user as fast-busy and is familiar to anyone who operates a telephone from a PBX.

• Receiver off hook Indicates that the receiver has been off hook for an extended period without placing a call.

• No such number Indicates that a subscriber placed a call to a nonexistent number.

Trunk Connections

Before a telephone call terminates at its final destination, the call is routed through multiple switches. When a switch receives a call, it determines whether the destination telephone number is within a local switch or if the call needs to go through another switch to a remote destination. Trunks interconnect the telephone company and PBX switches, as shown in Figure 2-9.

The primary function of the trunk is to provide the path between switches. The switch must route the call to the correct trunk or telephone line. Although many different subscribers share a trunk, only one subscriber uses it at any given time. As telephone calls end, they release trunks and make them available to the switch for subsequent calls. There can be several trunks between two switches.

The following are examples of the more common trunk types:

• Private trunk lines (tie-lines) Companies with multiple PBXs often connect them with tie trunk lines. Generally, tie trunk lines serve as dedicated circuits that connect PBXs. On a monthly basis, subscribers lease trunks from the telephone company to avoid the expense of using telephone lines on a per-extension basis. These types of connections, known as tie-lines, typically use special interfaces called recEive and transMit, or E&M interfaces.

• CO trunks A CO trunk serves as a direct connection between a PBX and the local CO that routes calls; for example, the connection from a private office network to the public switched telephone network (PSTN). When users dial 9, they are connecting through their PBX to the CO trunk to access the PSTN. CO trunks typically use Foreign Exchange Office interfaces. Certain specialized CO trunks are frequently used on the telephony network. A direct inward dial trunk, for example, allows outside callers to reach specific internal destinations without having to be connected via an operator.

• Interoffice trunks An interoffice trunk is a circuit that connects two local telephone company COs.

• Foreign exchange (FX) trunks FX trunks are interfaces that are connected to switches supporting connections to either office equipment or station equipment. Office equipment includes other switches (to extend the connection) and Cisco devices. Station equipment includes telephones, fax machines, and modems.

Trunk Signaling

Lines and trunks must adhere to signaling standards just as telephony networks and telephone companies do. Trunk signaling serves to initiate the connection between the switch and the network. There are five different types of trunk signaling, and each applies to different kinds of interfaces, such as FXS, FXO, and E&M:

• Loop-start signaling

• Ground-start signaling

• E&M wink-start signaling

• E&M immediate-start signaling

• E&M delay-start signaling

The following sections explain these signaling types.


Loop-Start Signaling

Loop-start signaling allows a user or the telephone company to seize a line or trunk when a subscriber is initiating a call. It is primarily used on local loops connecting to residences rather than on trunks interconnecting telephone switches.

A telephone connection exists in one of the following states, as illustrated in Figure 2-10:

• Idle (on hook)
• Telephone seizure (off hook)
• CO seizure (ringing)

A summary of the loop-start signaling process is as follows:

1. When the line is in the idle state, or on hook, the telephone or PBX opens the two-wire loop. The CO or FXS has battery on ring and ground on tip.

2. If a user lifts the handset off the cradle to place a call, the switch hook goes off hook and closes the loop (line seizure). The current can now flow through the telephone circuit. The CO or FXS module detects the current and returns a dial tone.

3. When the CO or FXS module detects an incoming call, it applies AC ring voltage superimposed over the 48 VDC battery, causing the ring generator to notify the recipient of a telephone call. When the telephone or PBX answers the call, thus closing the loop, the CO or FXS module removes the ring voltage.

Loop-start signaling is a poor solution for high-volume trunks because it leads to glare, which is the simultaneous seizure of the trunk from both ends. Glare occurs, for example, when you pick up your home telephone and find that someone is already at the other end.

Glare is not a significant problem at home. It is, however, a major problem when it occurs between switches at high-volume switching centers, such as long-distance carriers or large PBX systems.


Ground-Start Signaling

Ground-start signaling, illustrated in Figure 2-11, is a modification of loop-start signaling that corrects for the probability of glare. It solves the problem by providing current detection at both ends.

Although loop-start signaling works when you use your telephone at home, ground-start signaling is preferable when there are high-volume trunks involved at telephone switching centers. Because ground-start signaling uses a request or confirm switch at both ends of the interface, it is preferable over other signaling methods on high-usage trunks, such as FXOs. FXOs require implementation of answer supervision (reversal or absence of current) on the interface for the confirmation of on hook or off hook.


E&M Signaling

E&M signaling supports tie-line type facilities or signals between voice switches. Instead of superimposing both voice and signaling on the same wire, E&M uses separate paths, or leads, for each.

To call a remote office, your PBX must route a request for use of the trunk over its signal leads between the two sites. Your PBX makes the request by activating its M-lead. The other PBX detects the request when it detects current flowing on its E-lead. It then attaches a dial register to the trunk and your PBX, which sends the dialed digits. The remote PBX activates its M-lead to notify the local PBX that the call has been answered.

There are five types of E&M signaling: Type I, Type II, Type III, Type IV, and Type V. The E&M leads operate differently with each wiring scheme, as shown in Table 2-1 and Table 2-2. Keep in mind that any of the E&M supervisory signaling types (that is, wink-start, immediate-start, and delay-start) can operate over any of the following wiring schemes.

Analog and Digital Voice Connections

Analog Voice Fundamentals

Interfacing Cisco Systems equipment with traditional analog telephony devices requires an understanding of the various interfaces used in the industry. This section introduces analog interfaces that you can select from, including Foreign Exchange Station (FXS), Foreign Exchange Office (FXO), and ear and mouth (E&M).


Local-Loop Connections

A subscriber home telephone connects to the telephone company central office (CO) via an electrical communication path called a local loop, as illustrated in Figure 2-1. The loop consists of a pair of twisted wires. One is called tip; the other is called ring, as shown in Figure 2-2.


The names tip and ring come from the plug used by the operators of yesteryear to interconnect calls. As you can see in Figure 2-2, the plug used by these operators resembles the plug you might use to connect your headphones to your home stereo equipment. There are three conductors on this plug. The conductor (that is, wire) connected to the tip of the plug is called the tip wire, and the conductor connected to the ring in the middle of the plug is called the ring wire.

In most arrangements, the ring wire ties to the negative side of a power source, called the battery, while the tip wire connects to the ground. When you take your telephone off hook, current flows around the loop, allowing dial tone to reach your handset. Your local loop, along with all others in your neighborhood, connects to the CO in a cable bundle, either buried underground or strung on poles.

Note
Your home telephone service is provided to you from your service provider by way of two wires. Your home telephone controls whether the service on these wires is activated via the switch hook inside the telephone.


Local-Loop Signaling

A subscriber and telephone company notify each other of the call status through audible tones and an exchange of electrical current. This exchange of information is called local-loop signaling. Local-loop signaling consists of supervisory signaling, address signaling, and informational signaling, each of which has its own characteristics and purpose. The three types of local-loop signaling appear on the local loop and serve to prompt the subscriber and the switch into a certain action.


Supervisory Signaling

Resting the handset on the telephone cradle opens the switch hook and prevents the circuit current from flowing through the telephone, as seen in Figure 2-3. Regardless of the signaling type, a circuit goes on hook when the handset is placed on the telephone cradle and the switch hook is toggled to an open state. When the telephone is in this position, only the ringer is active.


To place a call, a subscriber must lift the handset from the telephone cradle. Removing the handset from the cradle places the circuit off hook, as shown in Figure 2-4. The switch hook is then toggled to a closed state, causing circuit current to flow through the electrical loop. The current notifies the telephone company that someone is requesting to place a telephone call. When the telephone network senses the off-hook connection by the flow of current, it provides a signal in the form of the dial tone to indicate that it is ready.


When a subscriber makes a call, the telephone switch sends voltage to the ringer to notify the other subscriber of an inbound call, as illustrated in Figure 2-5. The telephone company also sends a ringback tone to the caller, alerting the caller that it is sending ringing voltage to the recipient telephone.


The pattern of the ring signal, or ring cadence, varies around the world. As depicted in Figure 2-6, the ring cadence (that is, ringing pattern) in the United States is 2 seconds of ringing followed by 4 seconds of silence. The United Kingdom uses a double ring of 0.4 seconds separated by 0.2 seconds of silence, followed by 2 seconds of silence.

Address Signaling

Although somewhat outdated, rotary-dial telephones are still in use and easily recognized by their large numeric dial-wheel. When placing a call, the subscriber spins the large numeric dial-wheel to send digits. These digits must be produced at a specific rate and within a certain level of tolerance. Each pulse consists of a "break" and a "make," as detailed in Figure 2-7. The break segment is the time that the circuit is open. The make segment is the time during which the circuit is closed. In the United States, the break-and-make cycle must correspond to a ratio of 60 percent break to 40 percent make.


A governor inside the dial controls the rate at which the digits are pulsed. The dial pulse signaling process occurs as follows:

1. When a subscriber calls someone by dialing a digit on the rotary dial, a spring winds.

2. the dial is released, the spring rotates the dial back to its original position.


3. While the spring rotates the dial back to its original position, a cam-driven switch opens and closes the connection to the telephone company. The number of consecutive opens and closes (that is, breaks and makes) represents the dialed digit.


A more modern approach to address signaling is touch-tone dialing. Users who have a touch-tone pad or a push-button telephone must push the keypad buttons to place a call, rather than rotating a dial as they did with pulse dialing. Each button on the keypad is associated with a set of high and low frequencies. Each row of keys on the keypad is identified by a low-frequency tone; each column of keys on the keypad is identified by a high-frequency tone. The combination of both tones notifies the telephone company of the number being called, hence the term dual-tone multifrequency (DTMF). Figure 2-8 illustrates the combination of tones generated for each button on the keypad.

IP Telephony Applications

Types of Deployment

When deploying VoIP technologies, design decisions should take into account the environment in which VoIP is being installed. This section considers three typical environments: the campus LAN, enterprise, and service provider environments.


Campus LAN Environment

Campus LAN environments, an example of which is illustrated in Figure 1-24, have grown tremendously in the past several years due to the demand for networked resources, instant business communication, and VoIP applications.


Components for integrated voice and data campus networks, as discussed previously in the "Packet Telephony Components" section, include the following:

• IP Phone

• Gateway

• MCU

• Application server

When you are designing the campus infrastructure for voice, you must consider the following key issues:

• Robust, fault-tolerant, highly available network design

• Ability to power IP phones

• Redundant power supply for network components

• Ease of IP addressing

• QoS enhancements

Cisco Systems' internal telephone network in San Jose can be considered a campus LAN environment. All desktop phones connect to Ethernet switches and are controlled by Unified CallManager applications. Unified CallManager also controls the gateways and other application servers, such as the Unity server.


Enterprise Environment

Enterprise networks grow and evolve as company services and locations change and expand. Heavy reliance on information processing and universal access to corporate information has driven network designs to provide reliable access, redundancy, reachability, and manageability. These same principles apply to designing corporation-wide voice access in the enterprise environment.

Enterprise networks can be either centralized or distributed call processing environments. In the centralized call processing environment, all of the components of the voice system are controlled by a single centralized call agent, such as Unified CallManager, regardless of their physical location. In a distributed call processing environment, the components of the voice network at each location can act independently.

Figure 1-25 depicts an enterprise centralized call processing environment. Centralized voice networks provide enterprise-wide voice access for calls and voice services controlled from a central site. In this environment, the central site provisions all voice services, such as Cisco Unified CallManager, voice mail, and unified messaging. IP phones at remote sites connect to Cisco Unified CallManager through the IP WAN for call processing.


Components for centralized voice enterprise networks include the following:

• IP phone

• Cisco Unified CallManager cluster (central site only)

• Gateway (all sites)

• MCU (central site only)

• Application server (central site only)

• SRST based on Cisco IOS software (remote sites only)

• IP WAN

Figure 1-26 shows an enterprise distributed call processing environment. Distributed voice networks place voice components at each site and utilize the WAN for intersite calls only.


Components for distributed voice enterprise networks include the following:

• IP phone

• Cisco Unified CallManager cluster

• Gateway

• MCU

• Application server (all sites)

• IP WAN

Modern enterprise network applications include:

• E-business

• E-learning

• Customer care

• Unified messaging

• Videoconferencing

• Voice calls placed from web pages

Service Provider Environment

Service provider environments, an example of which is illustrated in Figure 1-27, add another level of complexity to the voice environment. To be competitive, service providers must provide their business customers with more efficient, less expensive alternatives to the PSTN for voice and data services.


Requirements in the service provider arena include:

• Carrier class performance Voice gateways must provide service that minimizes latency and controls jitter. This level of performance allows customers to maintain voice quality as they migrate from circuit-switched voice to IP-based services.

• Scalability Design must accommodate rapid growth to enable service providers to grow with their customer base. An important aspect of scalability is the automation, configuration, and administration of IP networks and gateways for seamless expansion.

• Comprehensive call records supporting flexible service pricing This is the ability to extract IP session and transaction information from multiple network devices and from all layers of the network, in real time, to produce detailed billing records.

• Signaling System 7 (SS7) interconnect capabilities Tariffs favor interconnection using SS7 signaling because Inter-Machine Trunks (IMTs) are less expensive than ISDN-based facilities. This financial benefit equates to lower monthly expenses, reduced cost of goods that are sold, and higher margins for service providers.

IP Telephony Applications

As customers migrate their voice networks, they face a myriad of choices regarding interface types, components, and topologies. A good network design incorporates solutions for current requirements and allows room for future growth. It is important to understand how voice interfaces with a network and how the components fit together to provide service in any environment.


Analog Interfaces

A Foreign Exchange Station (FXS) interface, as depicted in Figure 1-16, provides a direct connection to an analog telephone, a fax machine, or a similar device. From a telephone perspective, the FXS interface functions like a telephone switch (for example, a PBX); therefore, it must supply line power, ring voltage, and dial tone.


The FXS interface contains the coder-decoder (CODEC), which converts the spoken analog voice wave into a digital format for processing by the voice-enabled device.

The Foreign Exchange Office (FXO) interface, shown in Figure 1-17, allows an analog connection to be directed at the CO of a PSTN or to a station interface on a PBX. The switch recognizes the FXO interface as a telephone because the interface plugs directly into the line side of the switch. The FXO interface provides either pulse or DTMF digits for outbound dialing.


In PSTN terminology, an FXO-to-FXS connection is also referred to as a foreign exchange (FX) trunk. An FX trunk is a CO trunk that has access to a distant CO. Because this connection is FXS at one end and FXO at the other end, it acts as a long-distance extension of a local telephone line. In this instance, a local user can pick up the telephone and get a dial tone from a foreign city. Users in the foreign city can dial a local number and have the call connect to the user in the local city.

The E&M interface, shown in Figure 1-18, provides signaling for analog trunking. Analog trunk circuits connect automated systems (PBXs) and networks (COs). E&M signaling is also referred to as "ear and mouth," but its origin comes from the term "Earth and Magneto." Earth represents the electrical ground, and magneto represents the electromagnet used to generate tone.


E&M signaling defines a trunk-circuit side and a signaling-unit side for each connection, similar to the DCE and DTE reference types. The PBX is usually the trunk-circuit side, and the telco, CO, channel bank, or Cisco voice-enabled platform is the signaling-unit side.


Digital Interfaces

In a corporate environment with a large volume of voice traffic, connections to the PSTN and to PBXs are primarily digital. Examples of digital interfaces include T1, E1, and BRI interfaces.

T1 Interface

A T1 interface, as illustrated in Figure 1-19, is a form of digital connection that can simultaneously carry up to 24 conversations using two-wire pairs. When a T1 link operates in full-duplex mode, one wire pair sends and the other wire pair receives. The 24 channels are grouped together to form a frame. The frames are then grouped together into Super Frames (groups of 12 frames) or into Extended Superframes (groups of 24 frames).


The T1 interface carries either CAS or CCS. When a T1 interface uses CAS, the signaling robs a sampling bit for each channel to convey in band. When a T1 interface uses CCS, Q.931 signaling is used on a single channel, typically the last channel.

To configure CAS, you must specify the type of signaling that the robbed bits carry (for example, E&M Wink Start). This signaling must match the PSTN requirements or the PBX configuration. This is considered in-band signaling because the signal shares the same channel as the voice.

To configure CCS, you must configure the interface for PRI signaling. This level of configuration makes it possible to use channels 1 to 23 (called B channels) for voice traffic. Channel 24 (called the D channel) carries the Q.931 call control signaling for call setup, maintenance, and teardown. This type of signaling is considered out-of-band signaling because the Q.931 messages are sent in the D channel only.


E1 Interface

An E1 interface, shown in Figure 1-20, has 32 channels and simultaneously carries up to 30 conversations. The other two channels are used for framing and signaling. The 32 channels are grouped to form a frame. The frames are then grouped together into multiframes (groups of 16 frames). In Europe and Mexico, the E1 interface is most often used, while in the United States the T1 interface is most commonly used.


Although you can configure the E1 interface for either CAS or CCS, the most common usage is CCS.

When an E1 interface uses CAS, signaling travels out of band in the signaling channel but follows a strict association between the signal carried in the signaling channel and the channel to which the signaling is being applied. The signaling channel is channel 16.

In the first frame, channel 16 carries 4 bits of signaling for channel 1 and 4 bits of signaling for channel 17. In the second frame, channel 16 carries 4 bits of signaling for channel 2 and 4 bits for channel 18, and so on. This process makes it out-of-band CAS.

When an E1 interface uses CCS, Q.931 signaling is used on a single channel, typically channel 17. When configuring for CCS, configure the interface for PRI signaling. When E1 is configured for CCS, channel 16 carries Q.931 signaling messages only.


BRI Interface

Figure 1-21 depicts an Integrated Services Digital Network (ISDN) Basic Rate Interface (BRI). You can use a BRI to connect the PBX voice into the network. Used primarily in Europe for PBX connectivity, BRI provides a 16-kbps D channel for signaling and two 64-kbps B channels for voice. BRI uses Q.931 signaling in the D channel for call signaling.



IP Phones

Figure 1-22 depicts physical connection options for IP phones. The IP phone connects to the network through a Category 5 or better cable that has RJ-45 connectors. The power-enabled switch port or an external power supply provides power to an IP phone. The IP phone functions like other IP-capable devices sending IP packets to the IP network. Because these packets are carrying voice, you must consider both logical and physical configuration issues.

At the physical connection level, there are three options for connecting the IP phone:

• Single cable A single cable connects the telephone and the PC to the switch. Most enterprises install IP phones on their networks using a single cable for both the telephone and a PC. Reasons for using a single cable include ease of installation and cost savings on cabling infrastructure and wiring-closet switch ports.

• Multiple cables Separate cables connect the telephone and the PC to the switch. Users often connect the IP phone and PC using separate cables. This connection creates a physical separation between the voice and data networks.

• Multiple switches Separate cables connect the telephone and the PC to separate switches. With this option, IP phones are connected to separate switches in the wiring closet. By using this approach, you can avoid the cost of upgrading the current data switches and keep the voice and data networks completely separate.

Multiple switches are used to do the following:

• Provide inline power to IP phones without having to upgrade the data infrastructure

• Reduce the amount of Cisco IOS Catalyst software upgrades needed in the network

• Limit the spanning-tree configuration in the wiring-closet switches

The physical configuration for connecting an IP phone must address the following issues:

• Speed and duplex settings

• Inline power settings

The logical configuration for connecting an IP phone must address the following issues:

• IP addressing
  

• VLAN assignment

• Spanning tree

• Classification and queuing

Many Cisco IP phones, such as the 7970G shown in Figure 1-23, contain a three-port 10/100 switch. One port is an internal port that connects the voice electronics in the telephone. A second port connects a daisy-chained PC, and the third port uplinks to the Ethernet switch in the wiring closet.


If a computer is connected to an IP phone, data packets traveling to and from the computer, and to and from the phone, share the same physical link to the access layer switch and the same port on the access layer switch. This shared physical link has the following implications for the VLAN network configuration:

• Current VLANs may be configured on an IP subnet basis. However, additional IP addresses may not be available for assigning the telephone to the same subnet as the other devices that are connected to the same port.
  

• Data traffic that is supporting phones on the VLAN may reduce the quality of VoIP traffic.

You can resolve these issues by isolating the voice traffic on a separate VLAN for each of the ports connected to a telephone. The switch port configured for connecting a telephone would have separate VLANs configured to carry the following types of traffic:

• Voice traffic to and from the IP phone (auxiliary VLAN)

• Data traffic to and from the PC connected to the switch through the IP phone access port (native VLAN)

Fundamentals of Packet Telephony Networks

The increased efficiency of packet networks (for example, VoIP networks) and the ability to statistically multiplex voice traffic with data packets allows companies to maximize their return on investment (ROI) in data network infrastructures. Multiplexing voice traffic with data traffic reduces the number of costly circuits dedicated to servicing voice applications.

As demand for voice services expands, it is important to understand the different requirements of voice and data traffic. Previously, voice and data networks were separate and could not impact each other. Today, it is necessary to determine the protocols available to control voice calls and ensure that data flows are not negatively impacted.

This section delves into the benefits of packet telephony networks and provides an overview of basic packet telephony operations. Additionally, the fundamental components of packet networks are introduced. Finally, as a design consideration, this section considers the fragile nature of voice packets.


Packet Telephony Components

The basic components of a packet voice network, as shown in Figure 1-12, include the following:

• IP phones Provide IP voice to the desktop.

• Gatekeeper Provides Call Admission Control (CAC), bandwidth control and management, address translation, and call routing.

• Gateway Provides translation between VoIP and non-VoIP networks, such as the PSTN. It also provides physical access for local analog and digital voice devices, such as telephones, fax machines, key sets, and PBXs.

• Multipoint control unit (MCU) Provides real-time connectivity for participants in multiple locations to attend the same videoconference or meeting.

• Call agent Provides call control for IP phones, CAC, bandwidth control and management, and address translation. The call agent also serves as a repository for call routing information.

• Application servers Provide services such as voice mail, unified messaging, or call center support.

• 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. The user can view video streams and hear the audio that originates at a remote user station. Cisco targets its VT Advantage product at desktop videoconferencing applications.

Other components, such as software voice applications, interactive voice response (IVR) systems, and softphones, provide additional services to meet the needs of enterprise sites.


Call Control

Call control allows users to establish, maintain, and disconnect a voice flow across a network, as shown in Figure 1-13.


Although different protocols address call control in different ways, they all provide a common set of services. The following are the basic components of call control:

• Call setup Checks call-routing configuration to determine the destination of a call. The configuration specifies the bandwidth requirements for the call. When the bandwidth requirements are known, CAC determines if sufficient bandwidth is available to support the call. If bandwidth is available, call setup generates a setup message and sends it to the destination. If bandwidth is not available, call setup notifies the initiator by presenting a busy signal. Different call control protocols, such as H.323, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol (SIP), define different sets of messages to be exchanged during setup.

• Call maintenance Tracks packet count, packet loss, and interarrival jitter, or delay, when the call is set up. Information passes to the voice-enabled devices to determine if connection quality is good or if it has deteriorated to the point where the call should be dropped.

• Call teardown Notifies voice-enabled devices to free resources and make them available for the next call when either side terminates a call.

From a design perspective, you can set up call control in either a distributed or centralized architecture. The following sections describe both types.


Distributed Call Control

Distributed call control, an example of which is shown in Figure 1-13, offers an environment where call control is handled by multiple components in the network. This approach to call control is possible where the voice-capable device is configured to support call control directly. This is the case with a voice gateway when protocols, such as H.323 or SIP, are enabled on the device. In Figure 1-14, each location contains a Cisco Unified CallManager cluster. Each cluster is capable of handling call processing. Therefore, the topology shown demonstrates one example of distributed call control.


Distributed call control enables the gateway to perform the following procedure:

1. Recognize the request for service
2. Process dialed digits
3. Route the call
4. Supervise the call
5. Terminate the call


Centralized Call Control

Centralized call control, an example of which is illustrated in Figure 1-15, allows an external device (call agent) to handle the signaling and call processing, leaving the gateway to translate audio signals into voice packets after call setup. The call agent is responsible for all aspects of signaling, thus instructing the gateways to send specific signals at specific times. Also, the centralized call control model can leverage Cisco's Survivable Remote Site Telephony (SRST) feature to provide redundancy in the event of a WAN outage by having the voice-enabled router at the remote site perform basic call processing functions. In the figure, a Cisco Unified CallManager cluster located at the Headquarters location is in charge of call control. Therefore, the topology shown demonstrates an example of centralized call control.


When the call is set up, the following occur:

• The voice path runs directly between the two gateways and does not involve the call agent.

• When either side terminates the call, the call agent signals the gateways to release resources and wait for another call.

The use of centralized call control devices is beneficial in several ways:

• It centralizes the configuration for call routing and CAC. In a large voice environment, centralization can be extremely beneficial.

• The call agent is the only device that needs the intelligence to understand and participate in call control functions. These call control functions enable the customer to purchase less expensive voice-gateway devices and point to a single device to handle call control.

MGCP is one example of a centralized call control model.


Real-Time Versus Best-Effort Traffic

Voice and data can share the same medium. However, their traffic characteristics differ widely: Voice is real-time traffic and data is typically sent as best-effort traffic.

Traditional telephony networks were designed for real-time voice transmission, and therefore they cater to the need for a constant voice flow over the connection. Resources are reserved end to end on a per-call basis and are not released until the call is terminated. These resources guarantee that voice flows in an orderly manner. Good voice quality depends on the capacity of the network to deliver voice with guaranteed delay and timing.

Traditional data networks were designed for best-effort packet transmission. Packet telephony networks transmit with no guarantee of delivery, delay, or timing. Data handling is effective in this scenario because upper-layer protocols, such as TCP, provide for reliable, although untimely, packet transmission. TCP trades delay for reliability. Data can typically tolerate a certain amount of delay and is not affected by interpacket jitter.

A well-engineered, end-to-end network is required when converging delay-sensitive traffic, such as VoIP, with best-effort data traffic. Fine-tuning the network to adequately support VoIP involves a series of protocols and features to improve quality of service (QoS). Because the IP network is, by default, best effort, steps must be taken to ensure proper behavior of both the real-time and best-effort traffic. Packet telephony networks succeed, in large part, based on the QoS parameters that are implemented network-wide.

Multiplexing

A two-wire analog local loop typically carries one call at a time. To make better use of wiring facilities, different multiplexing techniques have been implemented to enable two-wire or four-wire connections to carry multiple conversations at the same time.

Time-division multiplexing (TDM) is used extensively in telephony networks to carry multiple conversations concurrently across a four-wire path, as shown in Figure 1-10. TDM involves simultaneously transmitting multiple separate voice signals over one communications medium by quickly interleaving pieces of each signal, one after another. Information from each data channel is allocated bandwidth based on preassigned timeslots, regardless of whether there is data to transmit.

Frequency-division multiplexing (FDM), as illustrated in Figure 1-11, involves carrying multiple voice signals by allocating an individual frequency range to each call. FDM is typically used in analog connections, although its functionality is similar to that of TDM in digital connections. FDM is used in cable or digital subscriber line (DSL) connections to allow the simultaneous use of multiple channels over the same wire.


If you have cable television service at your home, the television channels are all carried (and multiplexed) over a single pair of wires. This includes both the audio signals and the video signals. All the channels are present on the cable wires all the time. When you select the channel you want to watch, your set-top cable tuner determines which channel is sent to your television.

Call Signaling

Call signaling, in its most basic form, is the ability of a device to communicate a need for service to a network. The call-signaling process requires the network to detect a request for service and termination of service, send addressing information, and provide progress reports to the initiating party. This functionality corresponds to the three call-signaling types:

• Supervisory signaling

• Address signaling

• Informational signaling

A basic call setup, as illustrated in Figure 1-6, includes supervisory, address, and information signaling components. The supervisory signaling is used, for example, to detect that a phone went off hook. Address signaling occurs when a caller dials digits, and information signaling is represented by the dial tone heard by the caller.

This call setup can be broken down into three major steps. These steps include:

1. Local signaling: originating side The user signals the switch by going off hook and sending dialed digits through the local loop.

2. Network signaling The switch makes a routing decision and signals the next, or terminating, switch through the use of setup messages sent across a trunk.

3. Local signaling: terminating side The terminating switch signals the call recipient by sending ringing voltage through the local loop to the recipient telephone.


Supervisory Signaling

A subscriber and telephone company notify each other of call status with audible tones and an exchange of electrical current. This exchange of information is called supervisory signaling, as shown in Figure 1-7.


There are three different types of supervisory signaling:

• On hook When the handset rests on the cradle, the circuit is on hook. The switch prevents current from flowing through the telephone. Regardless of the signaling type, a circuit goes on hook when the handset is placed on the telephone cradle, and the switch hook is toggled to an open state. This prevents the current from flowing through the telephone. Only the ringer is active when the telephone is in this position.

• Off hook When the handset is removed from the telephone cradle, the circuit is off hook. The switch hook toggles to a closed state, causing circuit current to flow through the electrical loop. The current notifies the telephone company equipment that someone is requesting to place a telephone call. When the telephone network senses the off-hook connection by the flow of current, it provides a signal in the form of a dial tone to indicate that it is ready.

• Ringing When a subscriber makes a call, the telephone sends voltage to the ringer to notify the other subscriber of an inbound call. The telephone company also sends a ringback tone to the caller, alerting the caller that it is sending ringing voltage to the recipient telephone. Although the ringback tone sounds similar to ringing, it is a call-progress tone and not part of supervisory signaling.

The ringing pattern in the United States is 2 seconds of ringing tone followed by 4 seconds of silence. Europe uses a double ring followed by 2 seconds of silence.


Address Signaling

There are two types of telephones, as shown in Figure 1-8: a push-button (tone) telephone and a rotary-dial telephone.


These telephones use two different types of address signaling to notify the telephone company where a subscriber is calling:

• Dual-tone multifrequency (DTMF) Each button on the keypad of a touch-tone pad or push-button telephone is associated with a pair of high and low frequencies. On the keypad, each row of keys is identified by a low-frequency tone and each column is associated with a high-frequency tone. The combination of both tones notifies the telephone company of the number being called, thus the term dual-tone multifrequency (DTMF).

• Pulse The large numeric dial-wheel on a rotary-dial telephone spins to send digits to place a call. These digits must be produced at a specific rate and within a certain level of tolerance. Each pulse consists of a "break" and a "make," which are achieved by opening and closing the local loop circuit. The break segment is the time during which the circuit is open. The make segment is the time during which the circuit is closed. The break-and-make cycle must correspond to a ratio of 60 percent break to 40 percent make.

A governor inside the dial controls the rate at which the digits are pulsed. For example, when a subscriber calls someone by dialing a digit on the rotary dial, a spring winds. When the dial is released, the spring rotates the dial back to its original position. While the spring rotates the dial back to its original position, a cam-driven switch opens and closes the connection to the telephone company. The number of consecutive opens and closes, or breaks and makes, represents the dialed digit.


Information Signaling

Tone combinations indicate call progress and are used to notify subscribers of call status. Each combination of tones represents a different event in the call process. These events, whose frequencies and patterns are listed in Table 1-2, include the following:

• Dial tone Indicates that the telephone company is ready to receive digits from the user telephone.

• Busy Indicates that a call cannot be completed because the telephone at the remote end is already in use.

• Ringback (line or PBX) Indicates that the telephone company is attempting to complete a call on behalf of a subscriber.

• Congestion Indicates that congestion in the long-distance telephone network is preventing a telephone call from being processed.

• Reorder tone Indicates that all the local telephone circuits are busy, thus preventing a telephone call from being processed.

• Receiver off hook Indicates that a receiver has been off hook for an extended period of time without placing a call.

• No such number Indicates that a subscriber has placed a call to a nonexistent number.

A call placed from your residential telephone uses all three types of call signaling. When you lift the handset, a switch in your telephone closes to start current flow and notifies the telephone company that you want to make a call (supervisory signaling). The telephone company then sends dial tone to indicate that it is ready to receive your dialed digits (informational signaling). You then dial your digits by pressing numbers on the keypad (address signaling).


Digital versus Analog Connections

Supervisory, address, and informational signaling must be carried across both analog and digital connections. Depending on your connection to the network, you must configure specific signaling to match the type of signaling required by the service provider. Figure 1-9 illustrates digital and analog connections coexisting in the same network.

Digital PBX connections to the network are common in many countries. They may be T1 or E1 lines carrying channel associated signaling (CAS) or PRI lines using common channel signaling (CCS).

CAS is a signaling method that allows passing on-hook or off-hook status by setting bits that are associated with each specific voice channel. These bits are carried in band for T1 and out of band for E1.

An ISDN connection uses the D channel as the common channel to carry signaling messages for all other channels. CCS carries the signaling out of band, meaning that the signaling and the voice path do not share the same channel.

Analog interfaces require configuration of a specific signaling type to match the provider requirement. For interfaces that connect to the PSTN or to a telephone or similar edge device, the signaling is configured for either loop start or ground start, the functions of which are discussed in later. For analog trunk interfaces that connect two PBXs to each other (that is, E&M interfaces), or a PBX to a CO switch, the signaling is either wink- start, immediate-start, or delay-start, with the signaling type set to 1, 2, 3, 4, or 5.

Privately Owned Switches

In a corporate environment, where large numbers of staff need access to each other and the outside, individual telephone lines are not economically viable. A PBX is a smaller, privately owned version of the CO switches used by telephone companies, as illustrated in Figure 1-4.


Most businesses have a PBX telephone system, a key telephone system, or a Centrex service. Large offices with more than 50 telephones or handsets choose a PBX to connect users, both in-house and to the PSTN.

PBXs come in a variety of sizes, from 20 to 20,000 stations. The selection of a PBX is important to most companies, because a PBX has a typical life span of seven to ten years.

All PBXs offer a standard, basic set of calling features. Optional software provides additional capabilities.

A PBX connects to telephone handsets using line cards and to the local exchange using trunk cards.

A PBX has three major components:

• Terminal interface The terminal interface provides the connection between terminals and PBX features that reside in the control complex. Terminals can include telephone handsets, trunks, and lines. Common PBX features include dial tone and ringing.

• Switching network The switching network provides the transmission path between two or more terminals in a conversation. For example, two telephones within an office communicate over the switching network.

• Control complex The control complex provides the logic, memory, and processing for call setup, call supervision, and call disconnection.

PBX Installations

PBX switches are installed in large business campuses to relieve the public telephone company switches from having to switch local calls. When you call a coworker locally in your office campus, the PBX switches the call locally instead of having to rely on the public CO switch. The existence of PBX switches also limits the number of trunks needed to connect to the telephone company's CO switch. With a PBX installed, not every office desktop telephone needs its own trunk to the CO switch. Rather, the trunks are shared among all users.

Small organizations and branch offices often use a key telephone system, as shown in Figure 1-5, because a PBX offers functionality and extra features that they may not require. A key system offers small businesses distributed answering from any telephone, unlike the central answering position required for a PBX. Notice in Figure 1-5 that telephones interconnect to a key system via connector blocks, while trunks coming in from the local exchange interconnect to the key system via termination blocks.


Today, key telephone systems are either analog or digital and are microprocessor based. Key systems are typically used in offices with 30 to 40 users, but can be scaled to support over 100 users.

A key system has three major components:

• Key service unit A key service unit (KSU) holds the system switching components, power supply, intercom, line and station cards, and the system logic.

• System software System software provides the operating system and calling-feature software.

• Telephones (instruments or handsets) Telephones allow the user to choose a free line and dial out, usually by pressing a button on the telephone.

Larger companies use proprietary telephone networks with PBXs. In a key telephone system, each telephone has multiple lines that allow users to access outside lines to their CO. When a call comes into the company, a line or a key lights up on the telephone and indicates that a particular line is in use. Users can call another extension or let another person know where to pick up a call by using an intercom function, such as an overhead paging system or speakerphone.

Key telephone system functionality has evolved over time to include a class called hybrid telephone systems. The hybrid system adds many features that were previously available only in PBXs. There is no single definition of the functions and features that are classified as a hybrid system because all vendors provide a mix that they believe gives them a competitive advantage.

The main difference between a key telephone system and a hybrid telephone system is whether a single-line telephone can access a single CO local loop or trunk only (key telephone system) or whether the single-line telephone can access a pool of CO local loops or trunks (hybrid telephone system).

Introduction to Voice Technologies

Voice over IP (VoIP) is experiencing explosive growth. Many corporate environments have migrated, are actively migrating, or are researching the process of migrating to VoIP. Some long-distance providers are using VoIP to carry voice traffic, particularly on international calls. Companies, such as Vonage, offer VoIP service as a replacement for traditional telephony service in the home.

Migration is a process that involves gradually phasing out old components and replacing them with new ones. Many terms have been used to describe the technologies and applications for transporting voice in a converged packet network environment. When designing a converged network, it is necessary to clearly define all requirements and understand the various options that are available.

An important first step in designing a converged network is to understand the traditional telephony network and how it interfaces with voice components. You must know, from the start, how legacy voice equipment is connected and its possible migration paths.

The next step toward a good design is being knowledgeable about the components available for VoIP networks. You should be aware of the difference between voice and data flows within the network and the tools for controlling voice calls. Network requirements vary according to the size of the location. Knowing the difference between campus, enterprise, and service provider environments is crucial for choosing the right components and technologies.

This chapter provides an overview of the basic telephony functions and devices, including private branch exchanges (PBXs), switching functions, call signaling, and multiplexing techniques. It also reviews the basic components of the VoIP network and identifies the different requirements in campus, enterprise, and service provider environments. Together, these concepts and techniques provide a solid introduction to the VoIP arena.


Fundamentals of Telephony Networks

In traditional telephony networks, many components and processes are transparent to the customer. As you move from traditional telephony networks to converged voice and data networks, you must manage new components and processes to ensure seamless end-to-end call handling. To maintain acceptable service levels, you need to understand which devices you must now support and the processes that are necessary to ensure end-to-end call functionality.

Basic Components of Telephony Networks

A number of components must be in place for an end-to-end call to succeed. These components are listed here and shown in Figure 1-1:

• Edge devices

• Local loops

• Private or central office (CO) switches

• Trunks

Edge Devices

The two types of edge devices used in a telephony network include:

• Analog telephones Analog telephones are most common in home, small office/home office (SOHO), and small business environments. A direct connection to the public switched telephone network (PSTN) is usually made by using analog telephones. Proprietary analog telephones are occasionally used in conjunction with a PBX. These telephones provide additional functions such as speakerphone, volume control, PBX message-waiting indicator, call on hold, and personalized ringing.

• Digital telephones Digital telephones contain hardware to convert analog voice into a digitized stream. Larger corporate environments with PBXs generally use digital telephones. Digital telephones are typically proprietary, meaning that they work with the PBX or key system of that vendor only.

Local Loops

A local loop is the interface to the telephone company network. Typically, it is a single pair of wires that carry a single conversation. A home or small business may have multiple local loops.


Private or CO Switches

The CO switch terminates the local loop and handles signaling, digit collection, call routing, call setup, and call teardown.

A PBX switch is a privately owned switch located at the customer site. A PBX typically interfaces with other components to provide additional services, such as voice mail.


Trunks

The primary function of a trunk is to provide the path between two switches. There are several common trunk types, as shown in Figure 1-2, including the following:

• Tie trunk A dedicated circuit that connects PBXs directly

• CO trunk A direct connection between a local CO and a PBX

• Interoffice trunk A circuit that connects two local telephone company COs

The telephone installed in your home is considered an edge device because it terminates the service provided by your local telephone company. The local loop is the pair of wires that come to your house and provide residential telephone service. Trunks are the interconnections between telephone switches. They can be between private switches or telephone company switches.

CO Switches and Switching Systems

Figure 1-3 shows a typical CO switch environment. The CO switch terminates the local loop and makes the initial call-routing decision.


The call-routing function forwards the call to one of the following:

• Another end-user telephone, if it is connected to the same CO

• Another CO switch

• A tandem switch (that is, an intermediary switch between the source and destination switch)

The CO switch makes the telephone work with the following components:

• Battery The battery is the source of power to both the circuit and the telephone. It determines the status of the circuit. When the handset is lifted to let current flow, the telephone company provides the source that powers the circuit and the telephone. Because the telephone company powers the telephone from the CO, electrical power outages should not affect the basic telephone, also known as a POTS (plain old telephone service) phone.

• Current detector The current detector monitors the status of a circuit by detecting whether it is open or closed. Table 1-1 describes current flow in a typical telephone.

• Dial-tone generator When the digit register is ready, the dial-tone generator produces a dial tone to acknowledge the request for service.

• Dial register The digit register receives the dialed digits.

• Ring generator When the switch detects a call for a specific subscriber, the ring generator alerts the called party by sending a ring signal to that subscriber.

Some telephones on the market offer additional features that require a supplementary power source that the subscriber supplies; for example, cordless telephones. Some cordless telephones may lose functionality during a power outage.

When configuring a PBX connection to a CO switch, the signaling should match that of the CO switch. This configuration ensures that the switch and the PBX can detect on hook, off hook, and dialed digits coming from either direction.

Switching systems provide three primary functions:

• Call setup, routing, and teardown

• Call supervision

• Customer ID and telephone numbers

CO switches switch calls between locally terminated telephones. If a call recipient is not locally connected, the CO switch decides where to send the call based on its own call routing information, which is stored in a call-routing table. The call then travels over a trunk to another CO or to an intermediate switch that may belong to an inter-exchange carrier (IXC). Although intermediate switches do not provide dial tone, they act as hubs to connect other switches and provide interswitch call routing.

PSTN calls are traditionally circuit-switched, which guarantees end-to-end path and resources. Therefore, as the PSTN sends a call from one switch to another, the same resource is associated with the call until the call is terminated.

CO switches provide local service to residential telephones. The CO switch provides dial tone, indicating that the switch is ready to receive digits. When you dial your phone, the CO switch receives the digits, then routes your call. The call routing may involve more than one switch as the call progresses through the network.

Cisco Wireless Router and Switch Services

While the preceding devices are the most common that organizations use in constructing and maintaining a WLAN, Cisco offers other devices that can help your organization provide a robust, feature-rich wireless solution.


Cisco 3200 Series Wireless and Mobile Routers

To connect mobile networks to a wireless network, Cisco offers its Cisco 3200 Series wireless and mobile routers, shown in Figure 1-19. Contained in rugged enclosures and offering 802.11g functionality, these small devices (they are about as wide and long as a pen) can fit in vehicles or in outdoor locales. They offer the capability to transfer voice, data, and video across mobile wireless networks.


These routers are targeted at public safety, homeland security, defense agencies, and transportation agencies that need a durable router in a compact design that can be installed in vehicles.


Cisco Catalyst 6500 Series Switches

The Cisco Catalyst 6500 Series switches are a popular line of Cisco switches. In addition to serving wired clients, these switches can also be upgraded with a WLAN Services Module (WLSM). The WLSM is a key component of the Cisco SWAN architecture and enables fast, secure WLAN roaming within and across IP subnets. It also enhances WLAN security and smoothes out WLAN deployment and subsequent management.


Cisco Wireless 7920 IP Phone

Convergence is bandied about in the world of technology. Think of convergence as a techie's Swiss Army knife. We have cellular telephones that can play video games and MP3s players that can take pictures. Who knows what else they will be able to do in the coming years. Cisco is no stranger to the world of convergence. In the realm of wireless networks, one of the more compelling and useful devices is the Cisco Wireless 7920 IP phone. This telephone, shown in Figure 1-20, looks like a cellular telephone; however, it connects via the WLAN infrastructure (through an Aironet AP, for instance) then to the organization's gateway to allow VoIP telephone calls.

The phone uses the 802.11b protocol and Cisco CallManager. The product is ideally suited for environments in which users need telephony, but are constantly on the move and cannot be pinned down to a hardwired telephone. For example, hospitals, warehouses, universities, and retailers are ideally suited for these telephones.


Cisco Compatible Extensions (CCX)

As wireless technology has exploded in popularity, Cisco has seen the necessity for providing a mechanism through which third-party vendors can ensure compatibility among products. As a result, Cisco developed the CCX program.

Through the CCX program, WLAN vendors licensefree of chargeWLAN technology from Cisco. After that technology is implemented into the vendor's product, it is tested at an independent, third-party lab. If the product passes the testing procedures, the vendor is allowed to add a Cisco-compatible logo with the product, indicating that it not only works with Cisco equipment, but also takes advantage of advanced features. Intel is an example of this program in action. The company earned Cisco-compatible status with its Centrino mobile technology. This has been integrated in a number of laptop computers, such as Dell, Hewlett Packard, and Toshiba, among others.

The CCX program has been rolled out in three iterations. The requirements of CCX Version 2 build on the requirements of Version 1. For example, Version 1 of CCX security demands:

• WEP
• IEEE 802.11 and 802.1X
• Wi-Fi compliance
• Windows Hardware Quality Labs (WHQL)

Version 2 also requires WPA compliance.

From connection for two wireless clients working in ad hoc mode to a hospital nurse connected via a wireless IP phone; from an enterprise connecting its clients and office buildings in a MAN to police cars equipped with wireless routers, Cisco has a number of devices that enable a plethora of wireless networking functionality.

Version 3 includes EAP-FAST, wireless multi-media, CCKM for EAP-FAST, and single sign on.

Cisco Client Adapters

As we move down the wireless food chain, we come to the devices connecting users with the wireless network. Wireless adapters can be fitted to a multitude of devices client PCs, personal digital assistants, network printers, and so forth. Cisco offers four client adapters for its Aironet line:

• Cisco Aironet 350
• Cisco Aironet 802.11a, 802.11b, and 802.11g CardBus Wireless Client LAN Adapter
• Cisco Aironet 802.11a, 802.11b, and 802.11g Peripheral Component Interconnect (PCI) Wireless Client LAN Adapter
• Cisco Aironet 5 GHz 802.11a Adapter

The type of client adapter you use depends on what type of computer you need to connect. Laptops and other devices with Personal Computer Memory Card International Association (PCMCIA) and CardBus combination slots use a CardBus device. Desktop and tower-style PCs use PCI adapters.


Cisco Aironet 350


The entry-level model of Cisco client adapters is its venerable Aironet 350 adapter, shown in Figure 1-15. These adapters are designed as PCMCIA or PCI devices, which allow them to work with both desktop and laptop PCs.

These adapters can be used in either ad hoc (meaning two or more computers connect among themselves) or infrastructure environments (meaning the clients connect to a WLAN) and use the 802.11b protocol, which allows them to work in the 2.4-GHz band with a range up to 800 feet at 11 Mbps. Although this product operates at just 11 Mbps, it is compatible with 802.11g APs (although speed is limited to the adapter's top speed of 11 Mbps).


Cisco Aironet 802.11a, 802.11b, and 802.11g CardBus and PCI Wireless Client LAN Adapters

The Aironet 802.11 a/b/g CardBus and PCI Wireless Client LAN Adapters allow for a variety of uses and applications. Although they provide the same functionality, the difference between the CardBus and PCI devices is their physical design and construction. The CardBus devices are suited for laptops and tablet PCs, but the PCI device is meant for desktop PCs.

The CardBus device (shown in Figure 1-16) plugs into an open CardBus or combo CardBus/PCMCIA slot and the end sticks out an inch or so, allowing its internal antenna to communicate with the WLAN. The PCI device (shown in Figure 1-17) is a card that plugs into an open PCI slot on the PC. The card is connected to a small antenna that can be adjusted for best connectivity to the WLAN.


In spite of their physical differences, the devices offer the same functionality and are the most feature-rich and functional. They both offer:

• 802.11a coverage
• 802.11b coverage
• 802.11g coverage
• Dual mode 802.11a and 802.11g coverage
• Trimode 802.11a, 802.11b, and 802.11g coverage

These devices support Wi-Fi Protected Access (WPA) and WPA2. They also support 802.1X authentication, which includes LEAP, EAP-TLS, PEAP-GTC, EAP-FAST, and PEAP-MSCHAP V2.


Cisco Aironet 5 GHz 802.11a Adapter

The Cisco Aironet 5 GHz 802.11a Adapter serves clients that need access to a WLAN using 802.11a technology, shown in Figure 1-18. This device uses a CardBus form and is designed for use with APs, such as the Cisco Aironet 1200 Series 802.11a AP or the Aironet 1130AG AP.

Because the device offers 802.11a functionality, it operates at speeds of up to 54 Mbps in the 5-GHz band. Data rates can be reduced to extend the device's range.

The adapter uses the Cisco Wireless Security Suite, which offers the EAP framework for user-based authentication. It also supports a number of 802.1X authentication modes that include Cisco LEAP, EAP-TLS, PEAP, and EAP-SIM.