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.

Figure 2-1. Local Loops

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.

Figure 2-2. Tip and Ring

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.


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.

Figure 2-3. On Hook

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.

Figure 2-4. Off Hook

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.

Figure 2-5. Ringing

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.

Figure 2-6. Ring Cadences

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.

Figure 2-7. Pulse Dialing

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


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


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 (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.

Figure 2-8. Dual-Tone Multifrequency

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.

Figure 2-9. Trunks

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. The two FX trunk interfaces are:

    - Foreign Exchange Office (FXO) interfaces An FXO interface connects a PBX to another switch or Cisco device. The purpose of an FXO interface is to extend the telephony connection to a remote site; for example, if a user on a corporate PBX wanted a telephone installed at home instead of in the local office where the PBX is located, an FXO interface would be used. The FXO interface would connect to a Cisco voice router, which would serve to extend the connection to the user's home. This connection is an Off-Premises eXtension (OPX).

    - Foreign Exchange Station (FXS) interfaces An FXS interface connects station equipment: telephones, fax machines, and modems. A telephone connected directly to a switch or Cisco device requires an FXS interface. Because a home telephone connects directly to the telephone company CO switch, an FXS interface is used.


The service provided by local telephone companies for residential phones uses a foreign exchange interface, specifically FXS. This service is provided on two wires. The service is considered a station-side connection because the interface terminates with a telephone.

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)

Figure 2-10. Loop-Start Signaling

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


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.


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.


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.

Figure 2-11. Ground-Start Signaling

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.

Table 2-1. PBX to Intermediate Device

Signaling Type


On Hook

Off Hook




Battery(-48 VDC)




Battery(-48 VDC)




Battery(-48 VDC)









Table 2-2. Intermediate Device to PBX

Signaling Type


On Hook

Off Hook





















Four-wire E&M Type I signaling, shown in Figure 2-12, is actually a six-wire E&M signaling interface common in North America. One wire is the E-lead; the second wire is the M-lead, and the remaining two pairs of wires serve as the audio path. In this arrangement, the PBX supplies power, or battery, for both the M-leads and E-leads. This arrangement also requires that a common ground be connected between the PBX and the Cisco voice equipment.

Figure 2-12. E&M Type I

With the Type I interface, the Cisco voice equipment (tie-line equipment) generates the E signal to the PBX by grounding the E-lead. The PBX detects the E signal by sensing the increase in current through a resistive load. Similarly, the PBX generates the M signal by sourcing a current to the Cisco voice equipment (tie-line equipment), which detects it via a resistive load. The signal battery (SB) lead provides battery, while the signal ground (SG) lead provides ground.

Type V, as illustrated in Figure 2-13, is another six-wire E&M signaling type and the most common E&M signaling form outside of North America. In Type V, one wire is the E-lead and the other wire is the M-lead.

Figure 2-13. E&M Type V

Type V is a modified version of the Type I interface. In the Type V interface, the Cisco voice equipment (tie-line equipment) supplies battery for the M-lead while the PBX supplies battery for the E-lead. As in Type I, Type V requires that a common ground be connected between the PBX and the Cisco voice equipment.

Types II, III, and IV are eight-wire interfaces, where the eight wires include the voice path. One wire is the E-lead; the other wire is the M-lead. Two other wires are SG and SB. In Type II, SG and SB are the return paths for the E-lead and M-lead, respectively.

The Type II interface, depicted in Figure 2-14, exists for applications where a common ground between the PBX and the Cisco voice equipment (tie-line equipment) is not possible or practical. For example, the PBX is in one building on a campus and the Cisco equipment is in another. Because there is no common ground, each of the signals has its own return. For the E signal, the tie-line equipment permits the current to flow from the PBX; the current returns to the PBX SG lead or reference. Similarly, the PBX closes a path for the current to generate the M signal to the Cisco voice equipment (tie-line equipment) on the SB lead.

Figure 2-14. E&M Type II

Type III, as demonstrated in Figure 2-15, is useful for environments where the M-lead is likely to experience electrical interference and falsely signal its attached equipment. When idle, Type III latches the M-lead via an electrical relay to the SG lead. When the PBX activates the M-lead, it first delatches the SG lead via the relay and signals normally, as in Type II. Type III is not a common implementation.

Figure 2-15. E&M Type III

Type IV, shown in Figure 2-16, is a variation of Type II. In this arrangement, the battery source and ground are reversed on the SB and M wires (as compared to Type II). This means that both the SB and SG wires are grounded. Type IV signaling is symmetric and requires no common ground. Each side closes a current loop to signal, which detects the flow of current through a resistive load to indicate the presence of the signal. Cisco voice equipment does not support Type IV.

Figure 2-16. E&M Type IV

E&M Wink-Start Signaling

Tie trunks have bidirectional supervisory signaling that allows either end to initiate a trunk seizure. In this way, one PBX seizes the trunk, which then waits for an acknowledgment reply from the remote end. The local end must differentiate between a return acknowledgment and a remote-end request for service. Wink-start signaling, shown in Figure 2-17, is the most common E&M trunk seizure signal type.

Figure 2-17. Trunk Supervisory Signaling: Wink-Start

The following scenario summarizes the wink-start protocol event sequence:

  • The calling office seizes the line by activating its M-lead.

  • Instead of returning an off-hook acknowledgment immediately, the called switch allocates memory for use as a dial register, in the area of memory it uses to store incoming digits.

  • The called switch toggles its M-lead on and off for a specific time (usually 170 to 340 ms). (This on-hook/off-hook/on-hook sequence constitutes the wink.)

  • The calling switch receives the wink on its E-lead and forwards the digits to the remote end. DTMF tones are forwarded across the E&M link in the audio path, not on the M-lead.

  • The called party answers the telephone, and the called PBX raises its M-lead for the duration of the call.

If the timing of the returned wink is too short or impossible to detect, the trunk uses immediate-start, which the following section describes.

E&M Immediate-Start Signaling

Immediate-start signaling occurs occasionally if a PBX vendor implements wink-start, shown in Figure 2-18, but does not conform to the standards.

Figure 2-18. Trunk Supervisory Signaling: Immediate-Start

The following scenario summarizes the sequence of events for the immediate-start protocol:

  • The calling PBX seizes the line by activating its M-lead.

  • Instead of receiving an acknowledgment, the calling PBX waits a predetermined period (a minimum of 150 ms) and forwards the digits blindly. DTMF tones are forwarded across the E&M link in the audio path, not on the M-lead.

  • The called PBX acknowledges the calling PBX only after the called party answers the call by raising its M-lead.

E&M Delay-Start Signaling

Delay-start signaling, as depicted in Figure 2-19, is the original start protocol for E&M.

Figure 2-19. Trunk Supervisory Signaling: Delay-Start

Delay-start is used when all of the equipment is mechanical and requires time to process requests. The following scenario summarizes delay-start signaling:

  • When you place a call, your calling switch goes off hook by activating its M-lead.

  • The called switch acknowledges the request by activating its M-lead, and then rotates armatures and gears to reset its dial register to zero.

  • When the dial register at the called switch is in the ready state, the called switch deactivates its M-lead.

  • The calling switch then sends dialed digits. DTMF tones are forwarded across the E&M link in the audio path, not on the M-lead

  • When the called party answers, the called switch again activates its M-lead.


Although a local loop consists of two wires, when it reaches the switch, the connection changes to four wires with a two- to four-wire hybrid converter. Trunks then transport the signal across the network, as shown in Figure 2-20.

Figure 2-20. Two-Wire to Four-Wire Conversion and Echo

Telephone networks can experience two types of echo:

  • Acoustic echo Acoustic echo frequently occurs with speakerphones, when the received voice on the speaker excites the microphone and travels back to the speaker.

  • Electrical echo Electrical echo occurs when there is an electrical inconsistency in the telephony circuits. This electrical inconsistency is called an impedance mismatch.

If the lines have a good impedance match, the hybrid (that is, the two-wire to four-wire conversion circuit) is considered balanced, with little or no reflected energy. However, if the hybrid is inadequately balanced, and a portion of the transmit voice is reflected back toward the receive side, echo results.

Some form of echo is always present, as illustrated in Figure 2-21. However, echo can become a problem under the following conditions:

Figure 2-21. Echo Is Always Present

  • The magnitude or loudness of the echo is high.

  • The delay time between when you speak and when you hear your voice reflected is significant.

  • The listener hears the speaker twice.

The two components of echo are loudness and delay. Reducing either component reduces overall echo. When a user experiences delay, the conversation can get choppy, and the words of the participants sometimes overlap.


Echo tolerance varies. For most users, however, echo delay over 50 ms is problematic.

There are two ways to solve an echo problem in your telephone network:

  • Echo suppression

  • Echo cancellation

The echo suppressor, as depicted in Figure 2-22, works by transmitting speech in the forward direction and prohibiting audio in the return direction. The echo suppressor essentially breaks the return transmission path. This solution works sufficiently for voice transmission. However, for full-duplex modem connections, the action of the echo suppressor prevents communication. Therefore, when modems handshake, the answering modem returns a 2025 Hz tone to the calling modem, which serves to disable the echo suppressors along the transmission path.

Figure 2-22. Echo Suppression

Echo suppression has shortcomings in addressing certain echo conflict situations. Echo cancellation, a schematic of which is shown in Figure 2-23, is a more sophisticated method of eliminating echo.

Figure 2-23. Echo Cancellation

Rather than breaking or attenuating the return path (as in echo suppression), echo cancellation uses a special circuit to build a mathematical model of the transmitted speech pattern and subtracts it from the return path.


Echo cancellation applies the same technology that is used in audio headphones to cancel ambient noise. The headsets used by airline pilots, for example, feature a suppression circuit that cancels ambient noise so that the pilot hears only the audio from the headset. Any ambient noise from the cockpit is cancelled. This is the same technology used in echo cancellers.

Echo cancellation is the most common method of removing echo in the telephone network today and is used when necessary to adjust for echo on a Cisco device.


The echo canceller removes the echo from one end of the circuit only. If echo is an issue at both ends of the circuit, you must apply another echo canceller at the other end.

Cisco Voice over IP Cvoice (c) Authorized Self-study Guide
Cisco Voice over IP (CVoice) (Authorized Self-Study Guide) (2nd Edition)
ISBN: 1587052628
EAN: 2147483647
Year: 2006
Pages: 111
Authors: Kevin Wallace

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