Interpreting PBX Terminology

The world of telecommunications and PBX systems includes a vocabulary unique
unto itself.You may find that many of the words and acronyms are familiar and
common if your background is based in the data world. Nevertheless, there are a
number of new terms and concepts that need to be understood before tackling the
integration of voice and data systems. In addition, some acronyms have multiple
meanings depending on whether you’re discussing voice or data. For example, the
acronym CDP, to a Cisco router guru, likely means Cisco Discovery Protocol. In the
voice world this term refers to Coordinated Dial Plan.
So, what are the common PBX terms you may encounter? Well, the first
is a T-1.A T-1 circuit is capable of carrying up to 24 voice channels (DS-0),
depending on provisioning.The total available bandwidth is 1.544 Mbps, although
the Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI),
which uses T-1 framing, takes one DS-0 for upper layer signaling.The European
standard is called E-1. It provides, however, 2.048 Mbps of bandwidth, or 32 channels.
An E-1-based PRI, on the other hand, uses two of these channels for signaling
and framing, and thus, allows for 30 user-based voice channels. In addition
to the T-1 ISDN PRI , the circuit may also be configured as channel associated
signaling (CAS) or ear-and-mouth (E&M).
It is warranted to expand on ear-and-mouth technology slightly in this
forum, as E&M ports are found on the Cisco hardware platforms and many
interconnections will make use of this specification. E&M can also stand for earth
and magneto, amongst other variations, and is simply another signaling methodology.
E&M, like FXO and FXS, is an analog specification, unlike ISDN, which is
digital. In addition, FXO is available for PSTN or PBX connections, whereas
E&M is for trunk or tie lines between switches—they are network-to-network
links. As such, some Cisco installations use the VIC-2E/M interface for connections
to voice mail or legacy PBX systems. Please note that this module supports
both the two and four wire specifications of E&M for types I, II, III, and V.
These links may also be loop start, in which removing the receiver from the
hook closes a circuit and creates a loop, allowing connections. Or they may be
ground start, where an earth ground is needed to complete the loop and allow
connectivity.The term central office is a legacy description of the local telephone company’s
termination point for all numbers in a given area, and commonly connects to
PBXs via T-1s. Historically these were centrally located and copper was run from
each building in the town to the central office.Today, a wide variety of devices
are deployed to convert copper local loops into fiber and the central office terminates
a small number of fiber pairs that service hundreds of lines.The central
office would also provide a Direct Inward Dial (DID), although such connections
are typically bi-directional today. In order to directly connect from the public
phone system to a PBX, the caller must either be manually routed to the extension
or a relationship between the extension and a public number must be established.
DID provides the latter service—a block of numbers can be assigned to a
trunk line from the telephone provider to the PBX, and the PBX administrator
can route those numbers to related extensions. Figure 1.2 illustrates the logical
configuration of number 415-555-1234 to extension 51234. Please note that it is
quite common to create five-digit extensions in North America that relate to the
assigned DID numbers.To understand the routing in the phone network, one needs to understand
Coordinated Dial Plans (CDP). (As we mentioned earlier in this section, if you
are entering the world of telephony from the Cisco router, you are no doubt
thinking Cisco Discovery Protocol for CDP.The acronym CDP stands for both
actually, depending on your perspective.) A coordinated dial plan is analogous to
addressing in IP routing—the dial plan defines what numbers exist on your network
and how callers will reach phones outside your company. (For example, acoordinated dial plan may require a nine to be dialed before an external number.)
The term call routing has two meanings, however, that overlap slightly.The first
context is the physical act of routing a call through the network. For example,
calls to 312 are destined for Chicago, which is a long-distance call requiring a
service provider beyond the PBX.The second meaning involves the act of processing
that call—there may be three alternate paths to Chicago, and, based on
availability, price, and preference, an administrator can route the call along any of
those paths.In the voice world, telecommunications services are billed at various amounts
based on the tariff involved. A flat rate structure removes per-minute charges
from the billing calculation. Other tariffs can remove distance or other parameters
from the calculation.
It is also important to consider the historical import of which end is which in
the voice world. For example, you may hear the term tip-and-ring in single pair
copper connections, which relates to which end supplies the voltage on the wire.
In the same manner, there are also foreign exchanges, which have slightly different
meanings depending on your background.For our purposes, we will describe a Foreign Exchange Station (FXS) as a link
between the switch and an extension.This term is sometimes used to describe a
connection that services an analog device within the company attached to the
PBX, such as a fax machine or modem. If you’ve worked within the phone company,
this term may be defined differently; however, this definition is best in the
context of AVVID. In contrast, a Foreign Exchange Office (FXO) link is betweenthe PBX and the central office. It is a DS-0 and analog, and it is tariffed at a flatrate.
There may be instances when local services are desired, but ISDN or T-1
bandwidths are not needed. FXO connections can be used to service these situations,
and can also be used to provide local 911 service in the event all other calls
traverse the private network to a main site in another location.
Working with Analog Systems
Analog waves, unlike digital signaling, have a range of values that represent transmitted
information.These signals are susceptible to many forms of interference,
and, visually represented, they appear as a continuous wave. Figure 1.3 illustrates a
common analog waveform.As shown, there is no absolute value within the wave—it varies as the
strength of the signal increases or decreases.This introduces one of the primary
problems with analog systems, because one must consider the introduction of
static and amplification in the waveform.To illustrate this, consider Figure 1.4.
Note that the waveform is now comprised of higher highs and lower lows,
and spikes of noise have slightly altered the waveform.The receiver will perceive
this as a change in pitch, volume, and tone, and, should this degradation continue
through multiple amplifiers and noise-prone circuits, the original waveform may
be so disrupted that communications is impossible.The phone system was originally designed to make use of limited frequencies
to transmit voice signals. As human speech consumed a very small spectrum, the
analog telephone equipment could perform the relatively simple mechanical to
electrical conversion necessary to propagate a voice over long distances.
As with record players and compact disc/DVD players, it is likely that both
analog systems and their digital counterparts will remain for some time. As such,
it is important to consider how analog systems integrate into digital environments
such as VoIP or AVVID. Simply put, such installations will require conversions
from analog to digital, and, as with old 45s, the quality and performance of the
older systems may be limited. Of course, it will also be familiar and, at a political
level, you may find reluctance in getting users off their non-VoIP systems.
In the next section we will present digital systems. It needs to be noted here,
however, that there is a way to convert from analog to digital—a conversion
addressed by a coder-decoder (CODEC).The actual conversion is effectively a
sampling of the analog stream and a digital representation of that stream. Of
course, the conversion can take the digital data and interpolate an analog waveform.
The conversion is not without potential loss, unfortunately, and it is best to
limit the number of conversions within a data flow. Recall that FXO, FXS, and
E&M are all analog connection methods.
Benefiting from Digital Systems
Digital signals are binary in nature, and are either on or off.These states are very
precise, and unlike the continuous waveform that exists in analog systems, the
signal can be regenerated with accuracy regardless of noise and interference.This
is not to infer that digital signals are impervious to noise and static, but, rather,
these problems are easily detected in a digital system and can be compensated for.
This is made possible by the absolute values transmitted on the wire. Figure 1.5
illustrates a digital waveform.Digital systems in telephony can take advantage of this binary state and augment
communications with additional features that are not available in analog
systems, including compression.This allows speech to be sent in fewer bits than in
analog format, and, in the migration to AVVID, the data stream can actually be
stopped when a party stops speaking.This can greatly increase the volume of
connections that can concurrently occur in the network. ISDN PRI is one of the
most popular digital connections.
Providing Video Services
It is atypical to include the PBX as part of a video solution; however, some
advanced PBX systems do provide video services.These connections can either
be provided over broadband technologies or by way of Ethernet, but it is more
common in many systems to use the PBX as a termination point for multiple
ISDN Basic Rate Interface (BRI) channels.The BRI can transfer 128 Kbps of
user data, and these connections can be combined, or multiplexed, to provide
higher levels of bandwidth. Many video conferencing systems work well with
384 Kbps.
In later chapters, we will discuss the technical specifics of the various protocols
in use for these connections, including the H.320 specifications, which
govern the basic concepts regarding video transmission, including audio and
video processing, and are focused on lower-bandwidth media—ISDN and 56
Kbps specifically.This protocol supports point-to-point and multipoint sessions,
and provisioning for multicast or multipoint connections is an important consideration
in the video environment.
One of the first reactions many users have to compressed video is that it isn’t
like a television picture.The image is smaller and rougher, and, while it does not
have to be so degraded, most vendors haven’t forced the additional bandwidth or
processing requirements on end users. Adaptation, it is hoped, is to be driven by
function, which, in turn, may lead to faster networks and components.This will
likely be a slow process, as evidenced by the migration to high definition television
(HDTV).
In the United States, the analog video standard is called NTSC, or National
Television System Committee. Some in the industry claim that the acronym
should stand for Never Twice Same Color, being that, compared to the European
and Asian standards, the color information is poorly interpreted from set to set.
The NTSC standard specifies a frame rate, or screen refresh rate, of 30 framesper-
second (29.97). Users of these sets are quite accustomed to the grainy picture
provided and poor color resolution, and, while HDTV has been available in various
forms for years, the FCC and other authorities are already concerned in later
2001 that their 2006 mandate for HDTV conversion will fail.Video conferencing
may fail to generate sufficient drivers to make users upgrade their systems, and
may exist in degraded form for some time. Or it may also become the next
killer-application.This conundrum is a common theme in AVVID, and will be
interesting to watch as the old world meets the new.
Audio and video systems require common protocols to define the communications
stream, and these standards can be referred to as the H.300s, G.700s, and
the T.120s, in homage to the base numbering associated with each standard.This
is in addition to the transport protocols of ISDN, Digital Subscriber Line (DSL),Plain Old Telephone System (POTS), and others.The H, G, and T standards are
administered by the International Telecommunications Union (ITU).
The most universal of these video protocols is H.320, which defines a
number of parameters including picture size and bandwidth requirements, and
will operate within point-to-point and multipoint applications.
It would be unfair to only note H.320 in a discussion of video conferencing
protocols, however. H.261, for example, specifies the compression of real-time
audio and video data, and defines a screen size of 176 x 144 pixels (Quarter
Common Intermediate Format [QCIF]) to 352 x 288 (CIF). Most of these will
fit into the bandwidth availed by ISDN. H323 is most often referred to today, and
is commonly found in many applications, including the conferencing software
provided with Microsoft Windows.
The technicalities of all of these protocols is not important at this point in a
discussion of AVVID, and subsequent chapters will elaborate on the standards
used by Cisco’s CallManager and other resources, such as the IP phones.You will
find that many of the protocols used in AVVID telephony are the same as those
used in traditional video conferencing, and, because of this, there is integration
between the voice applications of the IP phone and the more traditional video
conferencing systems such as Microsoft’s NetMeeting. For example, one can call a
NetMeeting user from a Cisco IP phone.