What Voice Designers Do The art of voice system design is very different from data installations, although there are similarities.. A voice designer is typically confronted with two chal
Trang 1end 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
NOTE
In subsequent chapters, you will likely find that many of the problems encountered with VoIP systems, including latency, queuing, and routing, are related to the early decision of using IP as a protocol.
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
NOTE
Please note that this chapter is written from the Cisco AVVID perspective
as it relates to PBX systems and telephony, and, as such, some definitions and concepts will differ from the phone company or PBX system origins These are not errors, but rather, are simplifications of these terms and ideas to a common, related reference point For example, FXS and FXO in carrier terms can refer to other companies and their respective connections.
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
his-torically 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
Trang 2maintained 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 ser-vices provided by the key system were invaluable to companies of the mid-twen-tieth 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 pre-viously described, and consider those services in the VoIP environment
What Voice Designers Do
The art of voice system design is very different from data installations, although there are similarities A voice designer is typically confronted with two challenges—the tariff, or cost-per-minute-per-mile, and the redundancy within the network itself These designs are based on the number of channels needed, and are greatly simplified by the lack of routing protocols and intelligent end-stations.
For example, in a data network installation, the designer will typi-cally draw upon elements of the three-tier model This model defines a
network core, which interconnects different distribution layer devices,
and these, in turn, connect to the access layer, which services users This
design is based upon the concept that data packets will take alternate paths between devices based on load, in addition to the premise that the network devices themselves are prone to failure.
Voice designs are different in regards to both hierarchy and
redun-dancy First, the modern PBX is internally self-redundant, which means
the physical box itself attempts to provide its own redundancy Data net-working systems have only recently reached this level of redundancy, and, typically, they still experience a short outage as the system changes from the primary to standby engine In addition, the illusion of redun-dancy within the box in data networking often requires alterations to the connected devices—Cisco’s Hot Standby Router Protocol (HSRP) is a good example of how workstations are tricked into thinking that two
Designing & Planning…
Continued
Trang 3physically redundant routers are actually one device The trick is a shared
IP address and virtual Media Access Control (MAC) address to make two routers appear as a single router This, coupled with redundant Supervisor and routing engines, can create the appearance of a redun-dancy intradevice—however, because the end station has intelligence (unlike the phone), these installations are more complex.
As noted, the end stations in voice networks do not have intelli-gence, which greatly simplifies the redundancy model The internally self-redundant PBX, therefore, is not concerned with protocols and other user-side functions to provide redundancy Within the chassis, a PBX only needs to provide redundant power, redundant processing, and alternate egress paths Advanced systems may also provide an ability to redirect the physical port to another interface (engine) so the user’s phone is also serviced in the event of a hardware failure This is an uncommon installation, however Note that all of these redundancies occur intrachassis, and, because of the static nature of the switching paths, no convergence (compared to IP routing) occurs.
By now, you have likely guessed that the hierarchical design con-siderations in many PBX systems are also very different from routers and switches For example, it is rare to have a PBX system with three tiers Most large installations are serviced with two tiers sufficiently These designs parallel hub-and-spoke data models much more than the three-tier requirements of large data networks Part of this variance in design
is availed by the constant bit-rate of voice and the use of time division multiplexing (TDM) Thus, a designer in a PBX environment need only concern himself or herself with the number of concurrent calls between points All traffic consumes the same amount of bandwidth (a DS-0 in most cases).
Let’s look at that another way A data designer reviewing the capacity of a link needs two variables—the number of flows and the size
of each flow This is analogous to a freeway where semi trailers use the same road as cars and motorcycles Clearly the roadway can service more motorcycles than trucks In contrast, the voice designer needs only one variable in addition to time—the number of flows All flows are exactly the same—in the highway example, they would all be Volkswagens Thus, a designer need only consider the number of flows that will occur at the same moment This may result in a peak of 12 calls
growth and bursts in call volume The voice designer then adds resiliency and redundancy to the design, in addition to tariffs, or pricing, to develop a network.
Trang 4Looking Inside the PBX
A PBX consists of hardware and software designed to emulate the public tele-phone 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
Figure 1.1The Basic Functions of a PBX
Extension Termination
Trunk Termination
PSTN
Other PBX Systems
System Logic / Call Processing
Switching
Trang 5Category-3 or Category-5, and two- or four-wire (single pair or two pair)
installa-tions 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
NOTE
Some trunks are called tie lines, which are simply trunks used for
connec-tions to another PBX In some instances these connecconnec-tions are only capable of carrying voice channels; additional functions are provided in others One example is Siemens’ CorNet, which can provide most intra-PBX services between inter-intra-PBX devices.
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 net-work), 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
Trang 6happen 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 limi-tations of the PBX forwarding system
NOTE
Most PBX phones are digitally connected to the PBX and do not send the actual DTMF tones from the phone to the switch Traditional analog phones and some PBX phones will send the actual tones to be inter-preted by the switch However, the call routing is still based on the numbers pressed and received, and the non-Signaling System 7 (SS7) signaling is either proprietary or DTMF.
As a designer, you may specify that long-distance calls (indicated with a 9, fol-lowed 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
Trang 7processor 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 interna-tional 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 han-dling 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,
Trang 8Belarus, from Denver to Tel Aviv to Mozyr.This dialing plan addresses two fac-tors: 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 pro-tocol 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-net-work, and SS7 is typically not used in private switch-to-switch communications
How PBX Installation Differs from Router Installation
Most readers of this book are likely entering into the world of PBX sys-tems from the data world In fact, many of you may never have installed
a PBX or voice system However, whether you approach data from a voice background or voice from a data background, the reality is that at
a high level the two differ less than you might imagine.
It would be inappropriate to enter into the commands and syntax
of PBX configuration here—for one, which system would we use as a reference? There are many PBX systems, each with different software versions and hardware options, and each revision of code introduces new commands and syntax This is not unlike an academic conversation
on router configuration—Cisco or Nortel, Multilayer Switch Feature Card (MSFC) or Route Switch Module (RSM) or Route Switch Processor
(RSP)-2 In fact, this is the first of the ways in which the systems parallel one another PBXs and routers both have their own unique features and commands based on the vendor and the version of code.
Configuring & Implementing…
Continued
Trang 9Interpreting 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
In the previous sidebar, “What Voice Designers Do,” we discussed the design and deployment considerations of a modern PBX We also saw the similarities between data systems and voice PBXs These simi-larities include redundancy, cost/performance, and design limitations PBX systems augment these similarities with a few distinct differences, including:
■ Power Electrical requirements in PBX systems are frequently
48 volt DC Data network devices are often 120 volt AC.
■ Wiring It is rare that a PBX system will require Category 5
cabling for connectivity, unlike Ethernet In addition, it is uncommon to terminate voice grade wiring on patch panels Rather, voice wiring uses punch-down blocks that hold each bare wire onto a clip Requirements such as maintaining twists and staying under 100 meters are not part of the typ-ical voice installation.
■ Dial Plan Unlike IP routers, voice systems rely on static
routing tables when forwarding calls Calls are routed based
on a match with the destination number—unlike data net-works, the source address is rarely used for call routing The static route map will define a preferred path, an alternate path, and, sometimes, tertiary paths for each number within the environment.
■ Circuits In the data world, most circuits are billed at a
flat-rate per month These charges can be distance insensitive (as
in the case in Frame Relay), or distance sensitive (common in leased line connections) In voice, it is common to use leased line connections and the associated tariffs, which can allow for significant savings when traffic is carried on alternative paths These paths may be the connection to the long-dis-tance provider, or may be a private leased line between PBX systems.
Trang 10meanings 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 chan-nels An E-1-based PRI, on the other hand, uses two of these channels for sig-naling 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
method-ology 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 connec-tions 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
NOTE
It is important to remember that voice services are based on time divi-sion multiplexing, or TDM This is the basis for most connections in the voice world, just as it is for T-1 signaling A DS-0 is a single voice digital channel of traditional voice bandwidth—8Hz at 8 bits per sample.
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