To help you understand voice technology, this section discusses telephone technology plus transmission, signaling, and addressing concepts.

A telephone typically consists of the following components:

• Handset containing a transmitter and receiver
• Switch hook, which is a lever that is depressed when the handset is resting in its cradle
• Two-wire to four-wire converter to provide conversion between the four-wire handset and the two-wire local loop
• Dialer (either rotary or touch-tone)
• Ringer

A typical telephone does the following:

• Requests service from the network
• Performs dialing functions
• Performs a notification function (it rings)
• Provides answer and disconnect supervision
• Converts outgoing speech to electrical signals, and vice versa
• Automatically adjusts to the supplied power

For a telephone call to be completed, several forms of signaling must occur:

• Access signaling
• Station loop signaling
• Address signaling

The purpose of signaling in a voice network is to establish a connection. You typically begin a phohe call by taking a phone off hook, which sends an access signal. The line is seized, a path established across the network, and on the other end, the call is acknowledged.

Although various types of signaling are used in telecommunications today, this tutorial covers only those that directly apply to Cisco voice implementations. Modules 1 and 2 cover the most common analog signaling techniques supported by the Cisco 3640—loop start, ground start, and E&M; digital signaling is covered later.

Access Signaling

The first type involves access signaling, which determines when a line is off hook or on hook. When the handset is on its cradle, the phone is referred to as being on hook. When a telephone is on hook, the two wires do not touch, so the circuit (loop) is open and no current flows. When the handset is out of its cradle, it goes off hook. The wires touch, closing the loop, allowing current to flow through the two conductors that connect the phone to the network, sending an "off-hook" signal to the switch. To place a call, the phone must be off hook. To receive a call it must be on hook.

When the phone is on hook, no current flows

When the phone is off hook, current flows through
the conductors, and a signal is sent to the network.

There are two common methods of providing the basic access signal: loop start and ground start.

• Loop start. This is the most common technique for access signaling in a standard Public Switched Telephone Network (PSTN). Most residential telephones are analog loop-start telephones, based upon the concept of the subscriber loop, or local loop. The loop is an electrical communications path consisting of two wires, one for transmitting and one for receiving voice signals. The two-wire circuit is still referred to as the tip and ring, with the tip being tied to the ground and the ring tied to the negative side of the battery. When the phone handset is picked up (goes off hook), this action closes the circuit, establishing a loop between the PBX and the phone. Current is drawn from the battery of the PBX, indicating a change in status. This change in status signals the current detector in the PBX to provide dial tone. An incoming call is signaled to the handset by a standard on/off pattern, which causes the telephone to ring.
  
  This process is illustrated in the following diagram:
  

The handset goes off hook and closes the circuit,
establishing a loop between the PBX and the phone.

• Ground start. Ground start is another access signaling method used on on trunk lines or tie lines between PBXs to indicate on-hook/off-hook status to the CO. In ground-start signaling, one side of the two-wire trunk (typically the ring in the tip and ring configuration) is momentarily grounded to create dial tone.

  The action of a ground start is illustrated in the following diagram:

On a ground-start circuit, the user’s telephone detects the
flow of current on the "tip" lead and knows that the PBX is ready.

• When a user tries to place a call by grounding the ring lead, the PBX at the telco detects the flow of current and grounds the tip lead to indicate the PBX is ready.
• The user’s telephone perceives the flow of current on its "tip" and knows that the PBX is ready.

 

The tip is tied to the ground; the ring is tied to the negative side of the battery.

As you can see, the seizure of the line requires the cooperation of both parties to the call. A failure on either side stops the progress of the call. Therefore, both parties terminate service upon disconnecting (either an open tip or ring lead). This setup averts the disconnect supervision problem that might occur on the loop-start circuit, when a given line can be released only by the party who originated the phone call. For this reason, PBXs work best on ground-start trunks.

In a normal loop-start circuit, when you pick up the handset, you hear a dial tone indicating that a circuit is ready. On a ground-start circuit, however, the equipment at the user’s end should sense the flow of electrical current on the "tip" lead and interpret that the PBX is ready, so a dial tone from a PBX is not necessary, and its presence is optional. This setup allows the network to indicate off-hook status, or seizure of an incoming call independent of the ringing signal.

When the trunk is seized simultaneously from both ends, the resulting condition is known as glare. Glare makes loop start a poor solution for high-volume trunks such as those found in a workplace. Ground start corrects for glare by providing current detection at both ends.

FXO and FXS Signaling

A foreign exchange (FX) is a term applied to a trunk that has access to a distant CO. FX trunk signaling can be provided over either analog or T1 links and which utilize either loop-start or ground-start off-hook signaling techniques.

• Foreign eXchange Station (FXS): Standard residential phone lines are configured for FXS signaling. An FXS interface can be used to connect basic devices such as phones, modems, and faxes and must provide voltage, ring generation, off-hook detection, and call progress indicators.
• Foreign eXchange Office (FXO): FXO signaling is used primarily to communicate with CO switching equipment or PBXs. Because an FXO port on a router communicates directly with the PSTN or a PBX, it requires that a dialtone, ring indication, and call progress indicators be provided to it.

The basic design of FXS and FXO is simple but can cause problems, including disconnect supervision, probably the most common one. This problem can be fixed by implementing either disconnect supervision or using ground start.

E&M Signaling

Another analog signaling technique, used mainly between PBXs or other network-to-network telephony switches, is known as E&M, which stands for "ear and mouth" (or for "recEive and transMit"). There are five E&M signaling types, as well as two different wiring methods. Cisco's VoIP implementation supports E&M types I, II, III, and V, using both two-wire and four-wire implementations.

Cisco 3600 series and other Cisco voice-capable routers have available interfaces to support FXS, FXO, and E&M signaling types. The FXS and FXO interfaces are explained in more detail later in this module; E&M interfaces and configuration are covered in the next module of this course.

Station Loop Signaling

When the PBX receives the off-hook signal, it responds with an audible signal indicating that it is ready for a call—the dial tone. This two-way exchange between the PBX and the telephone is known as station loop signaling.

Address Signaling

In response to the audible prompt of the dial tone, the caller can request connection to another telephone by transmitting the address (telephone number) of the requested telephone (sometimes referred to as the called party identification number) to the PBX. This is known as address signaling.

Telephones generally use two basic types of address signaling: pulse and tone.

• Pulse dial (rotary dialing)
  
  Rotary dial phones represent the digit being dialed by momentarily stopping the current flow when the user turns the circular dial. For example, the circuit is broken three times, which creates three pulses in the current, to dial the digit "3."
  
  In general, the repetition rate of the pulses must be between 8 and 11 pulses per second (pps). The percent of the time that a break state is maintained within a pulse must be approximately 61 percent of the pulse length. This setup is depicted in the following diagram.
  
  Pulse Dialing
  
  
  
A complete explanation of pulse dialing is beyond the scope of this tutorial. Refer to EIA standard EIA/TIA - 470 for the technical details about pulse dialing.


• Tone dial (dual tone multifrequency, or DTMF; the method used by pushbutton telephones).
  
  DTMF is the most commonly used signaling system today. The keypad on a pushbutton phone has 12 keys. Each key press generates both a low-frequency and a high-frequency tone (the Dual Tone) that is specific to each individual key. The tones are then picked up and interpreted by telephone switches. The tones were selected to easily pass through the phone network with minimum interaction with each other and little attenuation.
  
  The following table describes the DTMF tone pairs.
  
  DTMF Tone Pairs
  

  Frequency in Hertz (Hz)	1209 Hz	1336 Hz	1477 Hz	1633 Hz
  697 Hz	1	ABC 2	DEF 3	(FO) [A]
  770 Hz	GHI 4	JKL 5	MNO 6	(F) [B]
  852 Hz	PRS 7	TUV 8	WXY 9	(I) [C]
  852 Hz	*	OPER 0	#	(P) [D]

  
  
  For most DTMF systems, there are 12 possible signaling states. (Only the values in columns 1–3 are typically used. You can see that these correspond to the buttons on a standard pushbutton telephone. The fourth column contains additional values used by the U.S. Government's AUTOVON network to indicate call priority.)
  
  In the North American phone system, the pound sign (#) is often used to end the interdigit timeout period. Pressing the pound key allows calls to be processed faster if the digits are immediately followed by the "#" stop-dial signal. The pound key can also be used as the first digit in a dialing sequence to notify the switch of a call requiring some kind of special treatment, such as a data call being made over a voice line.
  
  For automatic dialing, the minimum digit cycle time is specified as 100 msec. The duration of the DTMF signal must be at least 50 msec,and the minimum interval between digits is specified as 45 msec. The maximum interval is 3 seconds.

There is another type of signaling called multifrequency signaling (MF), which is used primarily on pay telephones, trunk circuits, and in some CCITT signaling schemes. MF is not covered in this tutorial.

Call Progress Indicators

While you are placing a call, you also hear a variety of audible signals that indicate the status of the call at various points along its path—for example, the dial tone, a busy signal, a signal indicating no circuit is available (fast busy), and ringing of the called party's phone. These tones are called call progress indicators.

Call Progress Indicators (North America)

In countries outside North America, the call progress tones you hear while placing a call may sound quite different from the ones represented in the table. The call progress tones in this table and the sound samples that you hear in this tutorial are based on North American standards. The call progress tones could be configured differently so that the frequencies and on and off times will vary.

When the PBX receives the DTMF digits that indicate the number to call, the PBX decides how to route the call: to the local telco's CO, to an internal telephone network, or to another PBX via a tie line. To route the call to the telco, the PBX signals to seize a trunk to the CO. Depending on the facilities, the signaling could be analog or digital. If the signaling is analog, the PBX would use either FXO-to-FXS signaling or E&M signaling. If the signaling is digital, one of two different methods of signaling can be employed: channel associated signaling, (CAS—also known as robbed bit signaling) or common channel signaling, (CCS).

E&M is discussed in module 2 "Analog Voice Internetworking with E&M." CAS and CCS are covered in module 3 of this course, "Basic Analog-to-Digital Voice over IP."

Answer Supervision Signaling

Assuming the call can be established, signaling would then occur at the remote end of the network. The CO seizes a line to the PBX and forwards the digits. The PBX selects the appropriate station, and signals an alert. The call proceeds, and the switch generates ring voltage to the phone. When the phone detects the voltage, it rings. The caller also hears an audible ring through the receiver; this signal is the ringback signal that is generated by the switch.

To ensure proper call handling and voice-mail routing, answer supervision signaling is particularly important. For example, if a phone is not answered after a specified number of rings, a call can be rerouted to voice mail based upon the lack of an answer supervision signal.

The methods of signaling discussed here are not the only available methods. For a more detailed discussion of signaling, including descriptions of supervision, address, call-progress, and network management signaling, read Signaling, which is included with this tutorial. You can also search Cisco Connection Online (CCO) for more information on this and other voice-related topics.

Go on to Understanding Voice over IP.