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Simply Speaking
A discussion of telephony

Article Number: A011
Article Type: Application
Modified: April 23, 1999

SUMMARY

Understanding voice technology is easy when you boil down the details into simple components. FXS and FXO interfaces used in telephony and voice communications are prime examples. These two voice interfaces operate very much like other items you are already very familiar with. But if I told you what they were now, you wouldn't read any further.

This article will help you understand these concepts by describing the relationship between these interfaces and terms, and how they relate to the operations of Athena Access and it's voice capabilities.

MORE INFORMATION

Introduction

To provide basic telephone service in your home you need two components: a handset (this is the telephone) and a telephone line (as provided by your telephone company). Together, they perform the functions necessary to provide the telephone service we've all become accustomed to.

By understanding the functions provided by each of these two common pieces you will easily grasp the operations of the FXS and FXO interfaces available on the Athena Access.

The Access FXO interface emulates the operations of a handset, while the FXS interface emulates the operations of a telephone line (actually, the Central Office (CO) switch). This is simple enough, or is it?

It's not uncommon to see various voice product's interfaces labelled exactly opposite to those presented on the Access. Don't be surprised, but do be aware. Users of voice equipment can be easily confused when dealing with multiple vendors. To help you through this, the Access interfaces can be explained in terms of what operations they perform and what they connect to.

Voice Interfaces for Athena Access

Athena Access provides three basic analogue voice interfaces. They are:

1. FXO
2. FXS
3. E&M

We've already presented the basic relationship between two of these interfaces, but there's other ways of describing these interfaces that will provide a better understanding.

The table below summarizes the operations of the three voice interfaces of Access. The left-most column is a list of basic telephony processes that occur during the normal operation of a telephone call. Each column relating to an interface describes how that interfaces deals with the related call progress signalling. For example, an FXO interface initiates an out-going call by going off-hook.

Telephone Switching Devices

At the other end of your telephone, the wires connect to a device that has the ability to switch your call to the desired location. It sounds simple but there is a level of intelligence required in this device in order to make decisions based on the information you dial and other electronic signals. These devices are generally called telephone switches, central office switches, Key Systems, or Private Branch eXchange (PBX).

For clarity, these devices are defined further below.

Key Systems

Key System Units (KSU) are "dumb" telephone switches that physically switch the caller's handset to another handset or to a trunk line. The KSU has trunk lines that emulate handsets and normally connect to telephone lines from a PSTN. Similarly, the KSU provides connections (often called station ports) that acts like telephone lines for connections to handsets. It is often very useful to simplify the interfaces in a telephone system as being either a handset or a telephone line to understand how a system will work and the limitations/capabilities of a system.

The Access can connect to these KSU trunks to provide end-to-end voice switching using FXS ports. The FXS port on the Access will detect the KSU going off-hook (loop start) and send dial tone. (Remember that the KSU acts like a handset when connecting to a trunk line.) The DTMF tones are detected by the Access and the call is routed, based on the routing table information.

For incoming calls, the FXS port will ring the KSU. The KSU must be configured to direct the call to a defined extension. This could be a person or automated attendant to allow for final re-direction to the end point handset. From the call originator's perspective, the call process takes two steps: one number to get the destination KSU and one to get to the destination extension.

The KSU can also use a station port as an outgoing trunk. A KSU station port acts like a telephone line (it provides dial tone, rings, detects off-hook, detects dialed digits to/from an attached device). In this case, an Access FXO port is used which emulates a handset. The process starts by a caller selecting an extension (dialing 9 or selecting a private Long Distance line button). The FXO port detects the ring from the KSU and goes off-hook. The Access can be programmed to automatically generate a connection to a specified location or it can send dial tone to the user's handset and expect DTMF dial digits. Since the KSU emulates a telephone line in this instance, it has no means to indicate when the original caller hangs up (does not open the loop as a handset does by going on-hook). The call must be cleared by the remote destination. If the remote is also an FXO port, once a call is set up, it can never be cleared except by network operator intervention using the console port. For this reason, it is strongly urged to never use Access FXO ports for connections to a KSU.

Access FXO ports should only be used for connections to PSTN telephone lines.

Private Branch eXchange

PBXs provide more options to allow a network of PBXs (connecting PBXs together). Most systems support both FXS and FXO interfaces but only for limited applications where full control of the call process is not required or is easily controlled. The FXO to FXO problem of clearing calls still exists and can't be used in this environment as well. An FXS interface has better "call supervision" as it can accept calls and hang up calls but it lacks the means of passing called numbers from the attached device. For improved call supervision between PBXs, E&M interfaces are supported. These interfaces support bi-directional call control, which means that calls can be initiated by either PBX, disconnected by either end, and signaling information (digits and tones) passed in both directions. While the FXS and FXO interfaces are simply a single pair of wires, the E&M interface is at least two pairs of wires and often, four pairs of wires. For example E&M Type 2, 4-Wire, has one pair of wires for receive voice and digits, one pair for transmitting voice and digits, one pair for transmitting signaling control and one pair for receiving signaling control. The receiving pair of wires for signaling control are referred to as the E leads (where E is the short form for Ear). Similarly, the transmitting pair are referred to as the M leads (where M is the short form for Mouth).

E&M Type 2, 4-Wire is the recommended way of connecting small numbers of trunks between PBXs as it avoids electrical grounding mismatches between PBXs, provides bi-directional call control, and avoids 2-to-4-Wire Hybrid connections between PBXs. At the expense of a few extra wires, a more reliable connection is provided between PBXs.

The call process between PBXs starts when a user places a call that is routed to a destination not on the local PBX. The calling PBX raises its M lead. After a brief delay the destination PBX places a short pulse on the E lead to indicate its readiness to accept digits. This is called "Wink Start" operation. The calling PBX can then send DTMF tones on the transmit voice pair to indicate the extension number on the destination PBX. When the connection is made, the called PBX raises the E lead. Either side can break the call by dropping the E or M lead.

Access works best when implementing E&M interfaces to PBXs.

As far as the PBX is concerned, it is just connected to another PBX (Access emulates a PBX). The call process starts by the caller placing a call to a remote destination. The calling PBX receives the dialed digits and determines that the route to the destination is via an E&M port to the Access. The PBX may be programmed to retain all of the called digits, delete digits, add digits, or completely change the called number. The Access detects the PBXs M lead, winks the PBX and the digits are passed to the Access. A similar process takes over in the Access for routing the call and modifying the called number. The Access must determine the correct DLCI to use that connects to the destination PBX by cross referencing the dialed number in a routing table. The FRF.11 call set-up protocol is used between the two Access units to pass the call digits and set up matching compression algorithms in both Access units. After routing the call to a remote Access, the process is repeated again and the call passed to the receiving PBX, and ultimately, the destination extension. Either end can clear the call by the user hanging up the handset. The PBX detects the opening of the loop and indicates the loss of connection to the other PBX by lowering the E or M lead.

Another View

Taking a slightly different tack, the text below is a direct excerpt from the "The Phreaker's Handbook" explaining the basic operation of a telephone:

When your telephone is ON-HOOK, there is 48 volts of
DC across the tip and the ring. When the handset of a fone 
is lifted a few switches close which cause a loop to become
connected between you and the fone company, or OFF-HOOK. 
This is also known as the local loop. Once this happens, 
the DC current is able to flow through your fone with less 
resistance. This causes a relay to energize which causes 
other CO equipment to realize that you want service. 
Eventually, you will end up with a dial tone. This also 
causes the 48 VDC to drop down to around 12 VDC. The 
resistance of the loop also drops below the 2500 ohm level; 
FCC licensed telephone equipment must have an OFF-HOOK 
impedance of 600 ohms.

When your fone rings, the telco sends 90 volts of pulsing 
AC down the line at around 15-60 Hz, usually 20 Hz. In 
most cases, this causes a metal armature to be attracted 
alternately between two electromagnets; thus, the armature 
often ends up striking two bells of some sort, the ring 
you often hear when non-electronic fones receive a call. 
Today, these mechanical ringers can be replaced with more 
modern electronic bells and other annoying signaling 
devices, which also explains why deaf people can have lights 
and other equipment attached to their fones instead of 
ringers.

When you dial on a fone, there are two common types of 
dialing, pulse and DTMF. If you are like me, you probably 
don't like either and thought about using MF or blue box 
tones. Dialing rotary breaks and makes connections in the 
fone loop, and the telco uses this to signal to their 
equipment that you are placing a call. Since it is one fone 
that is disconnecting and reconnecting the fone line, if 
someone else picks up another fone on the same extension, 
both cannot make pulse fone calls until one hangs up. DTMF,
on the other hand, is a more modern piece of equipment and 
relies on tones generated by a keypad, which can be 
characterized by a 0,1,2,3,4,5,6,7,8,9/A,B,C,D on the 
keypad. Most fones don't have an A,B,C,D keypad, for these 
frequencies are used by the telco for test and other 
purposes.

Do you want to know what a "Phreaker" is?

A Final Note

When you understand these terms, interfaces and definitions, it allows you to appreciate the application of various analogue voice interfaces at a more understandable level.

• The Access FXS interface emulates a telephone line and would directly attach to a handset.

• The Access FXO interface emulates a telephone (or handset) and would directly attach to a telephone line.

• The Access E&M interface is the best interface to use to connect to another PBX.

With this in mind, you'll be on the proper path towards a successful voice application.

REFERENCES

If you're interested in a set of definitions for a variety of the terms presented in this document, try our Telephony Glossary.§

Copyright © 1999 Develcon Electronics Ltd.  All Rights Reserved.