The evolution of phone systems starts with the early experiments of Alexander
Graham Bell. In 1875, communication over long distances was handled by the
telegraph, a simple device that would transmit electrical pulses across a wire.
Though there is some dispute regarding Mr. Bell’s status as the inventor of the
basic telephone, he and his estate have successfully defended the original patent
on the invention. For our purposes, the transmission of sound over electricity is
what’s significant.
Early telephones were little more than extensions of this original discovery,
and party lines and local phone companies were quite common.The party line
placed a multipoint drop from the phone company to a number of homes, and
the operator would signal an incoming call by altering the ring frequency to a
custom signal (for example, one long and three short rings).These systems were
prone to the same issues as broadcast media today, especially eavesdropping.
By 1950, a human operator was needed in a more centralized fashion to service
the hundreds of phones that were installed. Human operators manually connected
calls at the physical layer by using huge switchboards, and call setup times
were very long.This system was a solid first-generation effort to link party lines
and private lines into a national network. However, many of today’s advanced
features, including conferencing, alternate billing, and voice mail were inconceivable
then.
In the last half of the twentieth century, phone technology made huge strides,
including analog switching, digital switching, trunking, and the first versions of
the modern PBX.The human operator was replaced with automated switches
that processed calls automatically, and corporations were able to provide privately
administered services that rivaled the phone company.You may recall that the
original public phone systems were virtually always installed and owned by the
government or by single corporations—a far cry from the divergent world of
today in many countries.
While the PBX remains an entrenched fixture in many organizations, like the
mainframe computer, it also gave life to the next generation of successors and
augmenters. In the mainframe world this is the personal computer, and in the
voice and PBX world this is the IP telephone. Many in the AVVID arena consider
the IP telephone a cornerstone, because it is the simplest of devices for the
end user to operate and because it integrates so well with the additional services
promised by the technology.Voice over IP (VoIP) is the common term used for
these systems. For completeness, and to simplify installation, IP was selected, as it
is the most common protocol in the current networking world.
Designing with Legacy Systems in Mind
Before you tackle the converged world of Cisco’s AVVID—even if you configure
PBX systems daily—it may be a good idea to read this chapter to renew your
understanding of what a PBX is and how it works.
However, before we enter the world of the PBX, there is a legacy system that
needs introduction.This is the key system.A key system is a multiline phone historically
found in offices with up to ten users. It is best thought of as those old,
clicking phones with the large, lit buttons.
It is possible to find such systems servicing up to 100 users, however, modern
economics and the lack of advanced features makes these installations less
common, and well-suited for replacement.
As contrasted with the PBX, these systems function by placing a single line
on more than one physical phone and, typically, a one-for-one relationship is
maintained between the number of phones and the number of outside lines. As
such, unlike the PBX, these systems do not scale to hundreds of users, nor do
they save circuit charges.
So, why do we introduce the key system before the PBX? Well, the key
system is to the PBX what, presumably, the PBX is to VoIP and AVVID.The services
provided by the key system were invaluable to companies of the mid-twentieth
century, as calls needed to be routed from one resource to another. In
addition, many PBXs today emulate the key system’s multiline presence, and this
service is available with the current offering of AVVID. As you read about the
internal functions of the PBX, consider the legacy of phone and key systems previously
described, and consider those services in the VoIP environment.
Looking Inside the PBX
A PBX consists of hardware and software designed to emulate the public telephone
system within a company, and provide paths into the Public Switched
Telephone Network (PSTN).These systems can be categorized into four primary
areas, with each area containing one or more functions:
Extension termination
Trunk termination
System logic and call processing
Switching
These functions are illustrated in Figure 1.1 and described in greater detail in
the next sections.
Implementing Extension Termination
Each resource on the private side of the PBX is commonly called an extension.
These devices have a direct, one-for-one connection to a port on the PBX.These
connections are typically digital, however, analog extensions for modems and
other services are available, and you will find that the term Foreign Exchange
Station (FXS) is commonly used for analog stations such as fax machines and
modems attached to the PBX (although this is an erroneous term). In addition,
there is a large population of PBXs attached, via analog links, to the extensions,
and while the current connections from many vendors are digital, there is
nothing wrong with the analog connections apart from the limitations of the
transport.Wiring for these connections is voice grade. However, it may include
Category-3 or Category-5, and two- or four-wire (single pair or two pair) installations
are common.The PBX must also provide these extensions with dial tone
generation, just as the public phone switch provides this service for non-PBX
attached phones.These interfaces also pass the Dual Tone Multi-Frequency
(DTMF) tones to the call processing engine that will be described shortly.
Implementing Trunk Termination
While not required, most PBX systems are connected to at least one T-1 circuit
for connectivity to either the PSTN or another PBX within the company.A
trunk is a T-1 or other type of circuit, which can carry multiple channels, or time
division multiplexed (TDM) data streams. Recall that these connections can carry
up to 24 voice connections depending on their framing and signaling. Please note
that trunks can also use the E-1 standard, which allows for 30 user channels.
Call Processing and System Logic
In addition to the user interface found on most PBX systems, there is also logic
that controls the flow of calls.The basic process is based on dialing plans, which
compare the DTMF tones to the route plans and paths configured on the PBX.
These tones represent the numeric values of the buttons, in addition to the
asterisk (*) and pound (#) keys. Using the phone number or extension dialed,
the PBX routes the call either to the external trunk (the link to the public network),
to another PBX within the company (which is carried on an internal
trunk), or to another extension within the PBX.This addressing is signaled using
the DTMF tones.
The PBX can also make decisions based on its static tables in a dynamic
fashion.You’re probably thinking this doesn’t make sense, but it does. Recall that
a PBX route plan specifies the path an outbound call should take.What would
happen if that path failed? Simply, the administrator would specify an alternate
path—analogous to a floating static route in Cisco routing.These less-preferred
routes could be configured for call overflow (where insufficient capacity exists on
the primary link) or trunk failure (where the link must completely fail before
taking an alternate path).This decision adds a dynamic to the typically static limitations
of the PBX forwarding system.
As a designer, you may specify that long-distance calls (indicated with a 9, followed
by a ten-digit number, for example) should use a trunk to long-distance
provider A, which also provides the lowest cost per minute to the company.The
alternate path, configured for overflow calls, might go to long-distance company
B, which may also charge more per call. A backup path, using the local exchange
carrier, may be configured in the event the first two paths are unusable.
The system logic and call processing functions typically include collections of
billing information and other call accounting data that can be used for capacity
planning and charge-back services.These functions are independent of the final
PBX functional area: switching.
Switching
In order to better understand the diversity of the call routing and circuit
switching processes, each is presented as a distinct element in this section. In
practice, you will likely find that the two are so inter-related as to be one. In
many systems, however, there is a difference.
Switching in the PBX system is the mapping of a channel on one interface to
another channel on another interface. For example, this may involve linking a
DS-0 to a DS-1 (T-1), or an FXS port to a T-1 trunk on another PBX.The logic
that decides which path to be taken is part of the call processing function. Once
established, however, the switching of these TDM packets is transparent to the
processor until the call is torn down.This is a significant difference between IP
networking and voice traffic, as a routing process typically takes place for each
packet—in voice, the call setup only requires processing before the call begins.
It is significant to note that, as with data networking switches, the technology
can be blocking or nonblocking and this, coupled with other factors, can greatly
impact total capacity. For example, Siemens’ blocking architecture can switch up
to 5,760 ports, while the nonblocking Intecom can switch up to 60,000 ports.
Establishing Links Outside the PBX
The systems outside the PBX are actually pretty simple once you understand the
internal systems.The voice world is made up of trunks, which interconnect public
or private switches.The basic functionality of these devices is no different for our
purposes.
However, there are a few things you should consider when thinking of
external PBX resources.These include the wide variety of phone numbers in the
international phone network, and the signaling protocol between switches in the
public network.
As you may know, calling internationally from your respective country can be
either a simple or difficult process.The administration of all the possible numbers is
also a daunting task. In either the legacy or AVVID environment, you’ll need to
work with these external-dialing plans to allow users to connect to other systems.
Consider your home telephone for a moment. In the United States, a call to
Israel would require calling 011 (the international escape code), 972 (the international
country code for Israel), 3 (the city code, similar to an area code), and the
local number, which may be six or seven digits. However, note that in some
countries, the city code may appear as 03. A call to Belarus would use a country
code of 375, and the city code and number may only contain five digits. A call
from another country to the US would require a three-digit area code and a
seven-digit number.As a PBX programmer, the system must be capable of handling
all the digits provided and routing the call to the correct destination.
Now, with the home phone, the routing of the call is simple—the phone
company takes care of it! But, when we enter the PBX, we may have multiple
paths to consider.Though this can become very complex, the basics might
involve the use of private links between systems (tie lines). Consider the United
States to Israel example again. It may be cheaper to route calls from Denver to Tel
Aviv through the private tie line terminating in Jerusalem rather than the public
network, and, although unlikely, it may be cheaper still to route calls for Mozyr,
Belarus, from Denver to Tel Aviv to Mozyr.This dialing plan addresses two factors:
call routing and call tariffing.
However, let’s presume our call to Mozyr is cheaper using the public network
and employing a link between New York and London. How does the network
understand our call and establish a path between Denver and Mozyr?
Well, this is the second point of external systems.The switches in the network
need to signal each other using a common protocol. In many networks, this protocol
is called Signaling System 7 (SS7).
Data network designers are probably used to in-band signaling, where the IP
address is part of each packet. No such mechanism exists in voice networks.
Rather, the signaling is out-of-band, or independent of the actual data. SS7 is
used between the switches to provide this dialog, and, in our call to Mozyr, the
Denver phone company switch might use SS7 to signal a path from Denver to
Chicago, and another link from Chicago to New York. Once the path is built
using SS7, a voice link is established and the call commences. Please note that this
does not occur with the PBX private connection to Jerusalem, as this is in-network,
and SS7 is typically not used in private switch-to-switch communications.
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