Fundamentals of PBX Private Networks

The basic elements of a PBX private network are PBX switch nodes and tie trunks. Tie trunks are telecommunications channels that directly connect two PBXs. The first analog transmission tie trunks were known as E&M interface signaling trunks. The term E&M originally comes from the works earth (earth grounding) and magnet (electromagnetic generated tones). The letters E&M have also become known as “ear” and “mouth” or “receive” and “transmit.” E&M supervision signaling (on-hook/off-hook signaling) is used for a variety of networking operations, but the most basic function is to pass address signaling (called party number) between two endpoint PBXs in the private network. The most basic private network consists of two PBXs and at least one E&M tie trunk.

During the mid-1980s, E&M signaling was incorporated into T1 circuit digital private line services. In the early 1990s, ISDN PRI services were the primary trunking services used for private network tie trunk operations. By the end of the decade, IP-based trunk services could support traditional E&M signaling capabilities for inter-PBX networking requirements.

The first large private networks consisting or three or more PBX systems were known as tandem tie trunk networks (TTTNs). Each PBX was assigned a location code, and each station was assigned a private network extension number. TTTNs were based on a nonhierarchical network of tie trunks. A station user at the originating switch would dial the location code of the destination switch and wait to receive another dial tone signal before dialing the desired extension number. The request for dial tone was carried from switch to switch over E&M tie trunks. The private lines also were used to provide the dial tone signal back to the calling station user.

The first smart ETNs were designed in the late 1970s. An ETN was an enhanced version of a TTTN. It was based on a hierarchy of tie trunks and PBX switching nodes, with multiple call routes between network endpoints. A major innovation was an automatic alternate routing (AAR) program that selected a call route based on the number dialed and the most economic (or preferred) route available at the time the call was placed. The tandem switch node in the early ETNs was equipped with routing tables for determining the best route for on-net calls, had the capability of modifying outpulsed digits (for rerouting and directing calls), and could allow or deny call routing privileges to certain station users. Switch nodes in the ETN one level below the tandem switches are known as main switches. Main switches have trunk circuits for local CO switching access, but all network calls must be routed through a directly linked tandem node. Switches working behind the main nodes are known as satellites and tributaries. These switches are equipped with tie trunks to the main node only. They lack local CO trunk services and attendant operator positions. CO and private network access is through the main switch for all off-premises incoming and outgoing calls.

PBXs with ETN options support the important features described in the next sections.

Uniform Dial Plan (UDP)

A UDP provides for a common multidigit (usually four or five) dial plan that can be shared across a group of networked PBXs for interswitch and intraswitch dialing. The UDP includes a network location code, comparable to a CO code, and a multidigit extension number. The UDP extension number is not necessarily the same number as the station directory number. The network location code is often designated as the RNX; the extension number as XXXX. The network location code determines how the call is routed. Whenever a UDP is used to route a call, the number it outputs is in the form of RNX-XXXX. This must be taken into account so that the correct digit deletion or insertion can be specified within the routing pattern, so that the receiving switch receives digits in the format it expects. The UDP software program automatically translates the RNX- XXXX number to the directory number associated with the called station for digit analysis and routing by the destination PBX system.

Automatic Alternate Routing (AAR)

AAR provides alternate routing choices for on-net calls carried over the private network. Based on a routing table designed by the customer, the AAR software automatically selects the most preferred (usually the least expensive) route over the hierarchical tie trunk network. If the first route in the program table is not available, another route may be selected if the station user calling privileges warrant a more costly route based on the individual’s FRL. Large PBX systems can support several hundred different routing patterns (originating and terminating nodal pairs), with more than a dozen different call routes per routing pattern. The AAR routing patterns are shared with the ARS feature.

AAR also provides for digit modification to allow on-net calls to route over nonprivate PSTN trunk facilities when an on-net call route is not available. Calls rerouted off the network require digit modification to translate an RNX-XXXX number to a direct distance dialing number for national or international calling. On-net calls that are routed off-net are then controlled by the ARS feature of the PBX system.

FRLs and Network Class of Service (NCOS)

FRL and NCOS are features that provide for multiple levels of restriction for users of the AAR and ARS features. FRLs and NCOS allow a certain call type to a specific station user and deny the call to another user. For example, some station users may be allowed to place calls only to private network nodes and not off-net long distance toll calls. A call type that is highly restrictive is international direct distance dialing. FRLs and NCOS are transparent to the station user and are assigned and programmed by the system administrator. Each system facility, station and trunk, is assigned an FRL. Whether or not a call is allowed is based on the originating caller’s FRL and the availability of idle trunk circuits within an assigned trunk group required to route the call. If the station user’s FRL is equal to or greater than the trunk group FRL, calls may be routed using a trunk circuit in the trunk group. If the station user’s FRL is less than the trunk group FRL, the call is denied. Most intermediate/large PBX systems support at least eight FRLs (0 to 7).

The NCOS level of the originating station user determines which call routes can be used to complete the call for a specific routing pattern while the call route is being established across the network. NCOS is also known as traveling class mark (TCM) because the assigned restriction level of the originating station user is embedded within the voice communications signal as the call is routed across the network. If trunk facilities at one tandem switch node are busy and an alternate route (more expensive) is available, the NCOS/TCM level determines whether or not the station user is allowed access to a different call route. A call can be blocked or denied anywhere in the network. Another private network feature, advanced routing (look ahead routing) can be used to avoid possible blocked calls at tandem switch nodes.

Automatic Circuit Assurance (ACA)

ACA is a feature that helps customers identify trunk errors and problems, particularly for private tie trunk circuits. The PBX system maintains a record of individual trunk performance relative to short and long holding-time calls. Holding time is the period from trunk access to trunk release. A system administrator defines short and long holding-time limits. When a possible trunk circuit failure is detected, the system automatically initiates a referral call to an attendant, station user with a display-based voice terminal, or system manager with a CRT monitor. Based on system measurements of holding times per call, referral calls may be placed to attendant, station, or system manager positions. The display information identifies the call as an ACA call, identifies the trunk group access code and the trunk group member number, and shows the reason for the referral (short or long holding time). The ACA feature provides better telecommunications service through the early detection of faulty trunk circuits and possibly reduces out-of-service time.

Virtual Private Networks (VPNs)

A VPN is based on a custom switched telecommunications service tariffed by an interexchange carrier, which permits a customer to establish a communications path between two stations using a UDP. PBX systems are linked to the carrier’s CO facilities using private tie trunks or through dial-up facilities using special access codes. The network facili- ties are shared as part of the PSTN. The key benefit of a VPN is a significant reduction in private line services between networked PBX systems. The first voice communications VPN service was AT&T’s SDN. SDN was designed to expand the scope of private network solutions to customers who could not justify dedicated private line services in support of their private networking requirements. Other similar services were soon available from AT&T’s competitors. VPN telecommunications services, from exchange carriers such as AT&T, MCI Worldcom, and Sprint, simplify private networking applications for business customers because:

1. The backbone private network carrier facilities are maintained and managed by the service provider.

2. The UDP and AAR/ARS databases are centralized in the service provider service control points.

3. Less training and specialized communications equipment is required to design, implement, operate, and maintain a private network.

4. Call accounting records and billing information are provided by the service provider for all on-net and off-net calls.

Most customers with private network requirements use a combination of traditional ETN tie trunk links (analog, mostly digital) and virtual network services for their on-net and off-net calling requirements. Private networks using VPN services can enjoy the following benefits:

• Flexible system and station user port capacity

• Integrated voice/data communications

• System compatibility across different PBX platforms

• Cost effective usage- and distance-sensitive pricing

• Porting of private network numbering plans

• National/international service transparency

• PSTN reliability and QoS standards

• Customized report options

• Centralized network management capabilities

• Secure access via screening provisions and enforced use of authorization codes

There are several options available for PBX access to the carrier network, including local exchange service access and special services access. Local exchange access is usually reserved for very low-traffic volume locations that cannot justify a special access service. Local exchange access is provided through a switched connection from a local exchange carrier’s equal access end office. A customer using this access method may presubscribe to the VPN carrier code, or individual station users can dial the carrier code.

Special access arrangements provide direct access to the VPN provider network facilities:

• PSTN analog trunk circuit facilities

• PSTN digital trunk circuit facilities

• Customer-provided access (local bypass)

Digital trunk and customer-provided access may support multiple exchange carrier services, not only VPN service.

The VPN service functions similarly to private line ETNs for on- and off-net calls. Basic call processing features include:

• Seven-digit dialing (national and international calls)

• Advanced numbering plan (flexible multidigit numbering plans)

• Private network interface to ETNs

• Flexible routing when conditions warrant

• Network intercept announcements for call completion procedures

• Network remote access from off-net locations

VPN call management features include:

• Authorization codes

• Originating call screening—Grouping of callers with same calling privileges

• Feature screening—Specifies calling privileges for each screened caller group

• Access line grouping call screening—Call restriction by specific off-net telephone numbers

• Partitioned database management—Partitioning of VPN locations into multiple autonomous network groups with direct distance dialing and private networks

Exchange carrier VPNs originated as a voice-focused service. During the past few years VPN services have evolved to support voice and data communications networking applications. Most of today’s VPN offerings are focused on packet switched services to support customer WAN data communications requirements, but can be used for voice networking needs. As VPN has evolved, PBX networking has evolved with the emergence of ToIP technology and the trend toward VoIP trunking.

Cabling System Fundamentals

A structured cabling architecture design is intended to accommodate telecommunications technology changes with minimal impact on any of the other cabling subsystems, such as, electrical cabling. The target life cycle of an average cabling installation is up to 20 years. It is expected that a few generations of telecommunications systems will be installed and replaced or upgraded. Another planning assumption is that networking and bandwidth requirements will certainly increase during the life cycle of the cabling system. The following are key factors used to specify networks and cabling, as identified by Avaya in its SYSTIMAX CSC guidebook:

• Usage patterns, including combined size and duration of peak loads for all applications

• Expected increase in bandwidth demands

• The number of users and anticipated changes in that number

• Location of users and maximum distances between them

• The likely rate of change in users’ locations (churn)

• Connectivity with current and future devices and software

• Space available for cable runs

• Total cost of ownership

• Regulations and safety requirements

• Importance of protection against loss of service and data theft

PBX systems traditionally have been based on a star network topology. A star network topology includes many point-to-point links radiating from central equipment. The early LAN topologies were based on ring-and-bus network designs. A ring network topology has a continuous transmission loop that interconnects every device. The most familiar example of a ring network topology is the IBM token ring LAN. A bus network topology is a communications link that connects devices along the length of a cable. The original Ethernet LAN was based on a bus network topology.

Today’s dominant LAN technology is based on Ethernet standards. The logical topology of an Ethernet LAN is a bus topology, but the physical topology of the network is a star. Ethernet workstations that connect to an Ethernet hub or switch communicate over a high-speed bus housed in a hub or switch, but these network nodes are connected in a clustered star network topology. The star topology favored by PBX systems and adapted by Ethernet LANs is now the accepted communications system network topology.

The first Ethernet LAN installations were based on coaxial cable used for the transmission medium. During the mid-1980s the cabling used by PBX systems, known as unshielded twisted pair (UTP), was adapted for Ethernet LANs. Telephony UTP cabling was classified by IBM’s cabling system specifications as Category 3 and was used for 10Base-T Ethernet LANs operating at 10 Mbps. A 10Base-T Ethernet LAN used two pairs of Category 3 UTP cabling. A 100Base-T4 Ethernet LAN used four-pair Category 3 UTP cabling. A 100-Mbps Fast Ethernet, also known as 100Base-TX, used two-pair Category 5 UTP cabling. The 1000-Mbps (1 Gbps) Ethernet, 1000BASE-T, uses four-pair Category 5 UTP cabling. The 1000Base-TX, a lower cost alternative to 1000Base-T, uses the recently introduced Category 6 UTP cabling. PBX system telephony requirements can be satisfied with any of these UTP cabling types, making possible a single network cabling system infrastructure for voice and data communications applications.

In the SYSTIMAX SCS guidebook, Avaya lists the following considerations for choosing the type of customer network cabling:

• Maximum distance between network hubs and nodes

• Space available in ducting and floor/ceiling cavities

• The levels of electromagnetic interference (EMI)

• Likely changes in equipment served by the system and the way it is used

• Level of resilience required

• The required life span of the network

• Restrictions on cable routing that dictate cable bend radius

• Existing cable installations with potential for reuse

For the past two decades, most customers have used or installed two different cabling systems for telephony and data LAN applications. The evolution of the PBX system to an IP telephony platform will allow the large installed base of customers with installed circuit switched PBX systems to slowly phase out an infrastructure with two cabling systems and allow customers who are designing an entirely new converged voice/data network the opportunity to install a single cabling system. PBX systems installed before 1990 were implemented with Category 3 UTP, but more recent installations may have been based on Category 5 UTP, the same wiring used for data LANs. A newly installed communications system installation likely would be based on a generic cabling infrastructure using Category 5 UTP to provide for future needs.

A generic cabling system is a structured telecommunications cabling system capable of supporting a wide range of customer applications. Generic cabling can be installed before the definition of required applications because application-specific hardware (telephones, computers, etc.) is not part of the structured cabling design. Generic cabling can be enhanced through the use of flood wiring, which is the installation of sufficient cabling and telecommunications outlets in a work area to maximize the flexibility of the location for devices connected to the network. Many customers are currently installing four or six telecommunications outlets per work area, although the recommended minimum is two.

Fundamental LAN Planning Guidelines

The Cisco guide recommends a detailed analysis of the following LAN elements:

• LAN/campus topology

• IP addressing plan

• Location of TFTP servers, DNS servers, DHCP servers, firewalls, network address translation (NAT) gateways, and port address translation (PAT) gateways

• Potential location of gateways and call telephony servers

• Protocol implementation including IP routing, Spanning Tree, VTP, IPX, and IBM protocols

• Device analysis including software versions, modules, ports, speeds, and interfaces

• Phone connection methodology (direct or daisy chain)

According to the Cisco guide, the significant LAN topology issues are:

• Available average bandwidth

• Available peak or burst bandwidth

• Resource issues can may affect performance including buffers, memory, CPU, and queue depth

• Network availability

• IP phone port availability

• Desktop/phone QoS between user and switch

• Network scalability with increased traffic, IP subnets, and features

• Back-up power capability

• LAN QoS functionality

• Convergence at Layers 2 and 3

IP addressing issues that should be reviewed are:

• Phone IP addressing plan

• Average user IP subnet size use for the campus

• Number of core routes

• IP route summary plan

• DHCP server plan (fixed and variable addressing)

• DNS naming conventions

Potential considerations with IP addressing include:

• Route scalability with IP phones

• IP subnet space allocation for phones

• DHCP functionality with secondary addressing

• IP subnet overlap

• Duplicate IP addressing

The locations (or potential locations) of servers and gateways are important to ensure that service availability is consistent across the LAN infrastructure and for multiple sites. Gateways and servers in the review should include:

• TFTP servers

• DNS servers

• DHCP servers

• Firewalls

• NAT or PAT gateways

• Call telephony server

• Gateway location

After determining the location of these network elements, the following issues should be analyzed:

• Network service availability

• Gateway support (in conjunction with the IP telephony solution)

• Available bandwidth and scalability

• Service diversity

IP telephony scalability and availability issues will be affected by protocols in the network. The following areas for the protocol implementation analysis are:

• IP routing including protocols, summarization methods, non-broadcast media access (NBMA) configurations, and routing protocol safeguards

• Spanning Tree configuration including domain sizes, root designation, uplink fast, backbone fast, and priorities in relation to default gateways

• HSRP configuration

• VTP and VLAN configuration

• IPX, DLSW, or other required protocol services, including configuration and resource usage

With regard to protocol implementation, the following issues should be reviewed:

• Protocol scalability

• Network availability

• Potential impact on IP telephony performance or availability

All network devices should be reviewed and analyzed to determine whether the network has the desired control plane resources, interface bandwidth, QoS functionality, and power management capabilities. The checklist for this process includes:

• Device (type and product ID)

• Software version(s)

• Quantity deployed

• Modules and redundancy

• Services configured

• User media and bandwidth

• Uplink media and bandwidth

• Switched versus shared media

• Users per uplink and uplink load sharing/redundancy

• Number of VLAN supported

• Subnet size, and devices per subnet

For establishing a network baseline, it is important that the following measurements be made to determine voice quality levels and potential problem areas:

• Device average and peak CPU

• Device average and peak memory

• Peak backplane use

• Average link use (prefer peak hour average for capacity planning)

• Peak link use (prefer 5 minute average or smaller interval)

• Peak queue depth

• Buffer failures

• Average and peak voice call response times (before IP telephony implementation)

Cabling questions that may help determine the readiness of the infrastructure for IP telephony readiness include:

• Does the building wiring conform to EIA/TIA-568A?

• Does your organization comply with National Electric Code for powering and grounding sensitive equipment?

• Does your organization comply with the more rigorous IEEE 1100-1992 standard for recommended practices of grounding and powering sensitive equipment?

• Does the organization have standards for data center and wiring closet power that include circuit distribution, available power validation, redundant power supply circuit diversity, and circuit identification?

• Does the organization use UPS and/or generator power in the data center, wiring closet, phone systems, and internetworking devices?

• Does the organization have processes to SNMP manage or periodically validate and test back-up power?

• Does your business experience frequent lightning strikes? Are there other potential natural disasters?

• Is the wiring to your building above ground?

• Is the wiring in your building above ground?

Network bandwidth consumption is required for each VoIP stream. In any conversation, two such streams are required: one in each direction. The required bandwidth per conversation will be based on several factors, but of primary importance is the codec used to digitize, compress, and convert an analog voice signal into IP format. The two codecs of most interest are G.711 and G.729A. G.711 is the TIA recommended codec to optimize IP telephony QoS because it reduces impairment of the voice signal across the network, but the signal is uncompressed and requires a high amount of bandwidth. To save on network transmission costs, G.729A is used for off-premises traffic because it uses a compression algorithm to reduce bandwidth requirements.

Network performance and capacity planning help to ensure that the network will consistently have available bandwidth for data and VoIP traffic and that the VoIP packets will consistently meet delay and jitter requirements. Cisco recommends the following six-step process for network capacity and performance planning:

1. Determine baseline existing network use and peak load requirements

2. Determine VoIP traffic overhead in required sections of the network based on busy hour estimates, gateway capacities, and/or CallManager capacities

3. Determine minimum bandwidth requirements

4. Determine the required design changes and QoS requirements based on IP telephony design recommendations and bandwidth requirements (overprovide where possible)

5. Validate baseline performance

6. Determine trunking capacity

PBX Circuit Switching Design | Fundamentals of PBX Circuit Switching

PBX circuit switched network designs differ between each manufacturer’s product portfolio and even among models within a portfolio. Although there are differences in the individual PBX system switch network designs, the main functional elements are the same. All port circuit interface cards transmit and receive communications signals via a directly connected TDM bus, but the time and talk slot capacities are likely to differ between systems. A very small or small PBX system switching network design may consist of a single TDM bus backplane connected to every port interface circuit card, but a larger PBX system with more than one TDM bus must be designed to provide connections between the TDM bus segments. The TDM bus connections may be direct connections or center stage switch connections. The center stage switching system complex may be based on a space switch matrix design using circuit switched connections or a broadband TDM bus interconnecting lower bandwidth TDM buses. Two of the leading suppliers of PBX systems, Avaya and Alcatel, also offer customers of their very large PBX system models a center stage ATM switching option that can also support switched LAN data communications applications.

Center Stage Switch Complex

The primary function of the center stage switching complex is to provide connections between the local TDM buses, which support port carrier interface transmission requirements across the internal switching network. Complex center stage switching systems may be used in PBX systems designed for 100 user stations, although the smaller systems typically have a single TDM bus design or multiple TDM buses with direct link connections between each bus. A center stage switching complex may consist of a single large switching network or interconnected switching networks.

A very small PBX system usually does not require a center stage switching complex because the entire switching network might consist of a single TDM bus. Individual TDM bus switch network designs require a TDM bus with sufficient bandwidth (talk slots) to support the typical communications needs of a fully configured system at maximum port capacity. Most small PBX systems based on a single TDM bus design can provide nonblocking access to the switch network at maximum port capacity levels. If the TDM bus has fewer talk slots than station and trunk ports, the switch network can still support the communications traffic requirements, if properly engineered.

There are a few small and intermediate PBX systems that have multiple TDM buses but no center stage switching complex. For example, an Avaya Definity G3si can support up to 2,400 stations and 400 trunks using three-port equipment cabinets, each with a dedicated TDM bus, but does not use a center stage switching complex to connect the TDM buses. PBX system designs like the Definity G3si use direct cabling connections between each TDM bus for intercabinet connections between ports. This type of design can support a limited number of TDM buses without a center stage switching complex, but more TDM buses require more direct connections between each bus. When the system design includes more than three TDM buses, the switch network connection requirements may become unwieldly and often very costly. During the 1980s the Rolm CBX II 9000 supported up to 15 port equipment nodes that required dedicated fiber optic cabling connections to link each cabinet’s TDM bus switching network because it lacked a center stage switching complex. A fully configured system required 105 direct link connections (fiber cabling, fiber interface cards), resulting in a very costly alternative to a center stage switching complex. Every new nodal addition to the system required new fiber optic connections to every existing cabinet node. The advantage of a center stage switching com- plex in an intermediate/large PBX system design is to simplify switch network connections between endpoints.

There are several center stage switch designs typically used in digital circuit switched PBXs:

1. Broadband (very large bandwidth) TDM bus

2. Single-stage switch matrix

3. Multistage switch matrix

Multiple TDM Bus Design | Fundamentals of PBX Circuit Switching

Most PBX systems have multiple TDM buses supporting the communications needs of the system ports. The individual TDM bus segments can be linked through a variety of methods based on the topology of the switching network design (see section below). Two user stations connected to port interface circuit cards housed in different port equipment cabinet frames would each be supported by different local TDM buses. A PBX switched network TDM bus may support a few port circuit cards (perhaps half a port carrier shelf), an entire port carrier shelf, or even multiple port carrier shelves within the same cabinet frame. Stackable single carrier cabinet designs sharing a common backplane for process- ing and switching functions also may be supported by a single TDM bus, but it is more than likely that port circuit interfaces housed in different cabinets will not share the same TDM bus. Based on these TDM bus segment scenarios, it is possible that a call between two user stations housed on the same port carrier shelf would require four talk slots per call and two station users on different port carrier shelves in the same equipment cabinet frame would require only two talk slots. The number of required talk slots will depend on the switch network design.

Whenever two communicating ports do not share a common TDM bus, the PBX processing system will assign the originating port a talk slot on its local TDM bus (the TDM bus directly connected to its port interface card) and a talk slot on the TDM bus that is local to the destination port. The destination port will likewise be assigned two talk slots: one on its local TDM bus and another on the originating port’s local TDM bus. This scenario requires at least four available talk slots divided between two TDM buses. Additional communications channels may be required to link the TDM buses, and the PBX processing system makes the necessary assignments per call. Multiparty conference calls across multiple TDM buses will require one talk slot per party per TDM bus used to complete the call connection. For example, a three-party conference call among three internal station users, each supported by a different TDM bus, may require nine talk slots: three talk slots per party (one per TDM bus) × three parties = nine talk slots (Figure 1).

Figure 1: Call across multiple TDM buses.

To minimize the number of inter-TDM bus connection requirements and increase the traffic handling capability of the PBX system, it is often recommended that groups of station and trunk circuit ports share a common TDM bus, instead of dedicating different TDM buses to different station or trunk interface port circuits. Station user groups with high intercom traffic requirements also should share common switch networking facilities to minimize inter-TDM bus connection requirements.

In some instances there will not be talk slots available on a local TDM bus when a station call is initiated. Voice-based communications systems traditionally have been designed to support more system ports than available local TDM bus talk slots. In a typical PBX system environment, it is rare that every station or trunk port will be active simultaneously, but there may be a blocked call if there are more provisioned station/trunk ports than total local TDM bus talk slots. PBX systems are designed with traffic engineering calculations to minimize blocked call attempts. Call blocking situations have a low probability of occurring if the system is correctly traffic engineered, but they may occur if there are more potentially active ports than available talk slots required to provide the circuit switched connection between the ports

Pulse Code Modulation | Fundamentals of PBX Circuit Switching

PCM is a sampling technique for digitizing the analog voice-originated audio signals. PCM samples the original analog signal 8,000 times a second. This is more commonly referred to as 8-KHz sampling. The sampling rate used to code voice audio signals is based on the frequency range of the original signal. To accurately represent an analog signal in digital format, it is necessary to use a sampling rate twice the maximum analog signal frequency, a calculation based on the Shannon theorem. The maximum frequency of human voice is about 3.1 KHz. This frequency was rounded up to 4 KHz for ease of engineering design, resulting in an 8-KHz (2 × 4 KHz) sampling rate for digitizing voice audio signals. An 8-KHz sampling rate translates into a one sample every 125 microseconds (8 KHz^–1; Figure 1).

Figure 1: PCM encoding.

Each digital sample is represented by an 8-bit word (2⁸ = 256 sample levels) that measures the amplitude of the signal. The amplitude of the signal is based on the power (expressed in units of voltage) of the electri- cal signal generated by the telephone transmitter/receiver in the handset. This signaling technique has become known as Digital Signaling 0 (DS0), or 64-Kbps (8 bits × 8 KHz) channel transmission format. The term DS0 was defined based on the Digital Signaling 1 (DS1) format used to describe a digital T1-carrier communications circuit supporting 24 64-Kbps communications channels.

The PCM samples generated from each communications system port are transmitted onto the TDM bus in a continuously rotating sequence based on the time slot assignments given to each port circuit interface (see below). Only a single PCM word sample is transmitted at a time; that is the entire electrical transmission line is reserved for use by only one port circuit for transmission of its sample signal. The PBX processing system monitors each port circuit’s transmission time assignment in the rotating sequence, controls when the sample is transmitted, and coordinates transmission of the sample between the originating and destination endpoints.

There are two standards for coding the signal sample level. The Mu-Law standard is used in North America and Japan, and the A-Law standard is used in most other countries throughout the world, although each uses the 64-Kbps transmission format. For this reason, PBX systems must be designed and programmed for different geographic markets. Using firmware downloads, system vendors and customers can program their PBX systems to support the local PCM standard. The early digital PBX systems used different hardware equipment based on the location of the installation.

To summarize the fundamentals of PCM:

1. 4-KHz analog voice signals are sampled 8,000 times per second (8-KHz sampling rate)

2. Each sample produces an 8-bit word number (e.g., 11100010)

3. 8-bit samples are transmitted onto the TDM bus at a 64-Kbps transmission rate

4. The samples from each port circuit are transmitted in a continuously rotating sequence

TDM Bus Bandwidth and Capacity

Bandwidth is the amount of data that can be transmitted in a fixed period. For digital transmission, bandwidth is expressed in bits per second; for analog transmission, bandwidth is expressed in cycles per second (Hertz). The bandwidth of an 8-bit PBX TDM bus is determined by the internal switching system clock rate used to create time slots for each channel’s transmission. The faster the clock rate, the more digitized samples per second can be transmitted over the TDM bus. The clock functions merely as a counter; the faster it “counts,” the more sampled digital signals within a fixed period (usually defined as 1 second) can be transmitted over the TDM bus. For example, an 8-bit TDM bus operating at 2.048 MHz has a bandwidth of 16 Mbps. If you double the clock rate (double the operating frequency), the bandwidth capacity doubles.

If the operating frequency of the TDM bus is not provided but the number of time slots is known, the bandwidth of a TDM bus can be calculated by multiplying the number of time slots (as determined by the system clock rate) by 64 Kbps (the number of transmitted bits per communications channel). A TDM bus segmented into 32 time slots has a transmission bandwidth of 2.048 Mbps (32 × 64 Kbps ). A system with a faster clock rate that is capable of segmenting the TDM bus into 512 time slots would have a bandwidth of 32.64 Mbps (512 time slots × 64 Kbps). It is usually awkward to refer to the TDM bus bandwidth by the exact transmission capacity, so it is common to see the TDM bus bandwidth written, and referred to, as a 32-Mbps TDM bus. The most common PBX TDM bus bandwidths are usually based on exponential multiples of 2 Mbps (2ⁿ Mbps): 2 Mbps, 8 Mbps, 16 Mbps, or 32 Mbps.

Not all of the time slot segments on a TDM bus are designed to handle communications traffic. Most PBX system TDM buses reserve a few time slots for the transmission of control signaling across the internal system processing elements. For example, a control signal time slot is used to alert the main system control complex that a telephone has gone off-hook. The signal is passed from the telephone instrument to the port circuit card and across the internal processing/switching transmission network via the local TDM bus. A variation of this design is to dedicate an entire TDM bus for control signaling. Examples of PBX systems using a dedicated signaling bus for system port control are the Siemens Hicom 300H and Hitachi HCX 5000 products. When a single bus is used for communications and processing functions, the control signaling time slots are not available for port communications requirements and should not be considered in the analysis of the system’s traffic handling capabilities. Time slots that can be used for real-time communications applications are sometimes referred to as talk slots. The total number of talk slots and control signaling slots per TDM bus are equal to the number of time slots (Figure 2).

Figure 2: TDM transmission bus.

The number of available talk slots limits the number of active PBX ports that can be simultaneously supported by a single, common TDM bus. An active PBX port is simply defined as a port that is transmitting and receiving real-time communications signals—on-line. A port may be customer premises equipment working behind the PBX system, such as a telephone, or an off-premises trunk circuit. For example, a 2.048-Mbps TDM bus with 30 talk slots can support 30 active communications ports (telephones, modems, facsimile terminals, trunk circuits, voice mail ports, etc.).

Port-to-Port Communications over a Single TDM Bus

When a station port is about to become active, the PBX processing system will assign that port a talk slot on the local TDM bus connected to the port’s circuit interface card. For the remainder of the call, the port will use the designated talk slot to transmit its digitized voice communications signals across the internal circuit switched network. The port receiving the call may be another station or a port interface connected to a trunk circuit. If the originating station port places an internal system call to another station, the PBX processing system will assign the destination port a designated talk slot on its interface circuit card TDM bus. If the originating station port is making an off-premises call requiring a trunk circuit connection, the processing system will assign the trunk circuit port a talk slot on the TDM bus supporting its interface circuit card. The same process takes place for incoming trunk calls to user stations: the trunk interface port circuit is assigned a talk slot, as is the destination station interface port circuit. The circuit switching system will use the two designated talk slots to connect the two ports together to transmit and receive communications signals for the duration of the call. The two talks slots will work in tandem for talking and listening between ports, with each port physically linked to both talk slots.

The number of talk slots required per call will depend on two conditions:

1. The number of connected ports per call

2. The number of TDM bus segments required for port connections

A two-party conversation between PBX ports interfacing with the same TDM bus will require two talk slots, but multiparty conference calls will require as many talk slots as conference parties. For example, a four-party conference call would require four TDM bus talk slots. A small PBX system based on a circuit switched network design consisting of a single TDM bus will require only two talk slots per two-party call, but intermediate and large PBX systems designed to support hundreds or thousands of station and trunk ports will have switching network designs based on many interconnected TDM bus segments, and more than two talk slots will be required for an internal two-party call. More than four talk slots will be required for a four-party conference call if the ports are housed in different port equipment cabinets. The answer to the question of how many talk slots are needed per call in an intermediate or large PBX system requires some knowledge of the PBX switch network design.

Time Division Multiplexing | Fundamentals of PBX Circuit Switching

The core design element of a traditional digital PBX is the local transmission bus that connects to a port circuit card. Many port circuit cards may share a common local transmission bus, and a PBX system may have many local buses dedicated to designated port circuit cards housed in different port carrier shelves and/or cabinets. Port circuit cards are used to connect peripheral equipment devices, such as telephones and telephone company trunk circuits, to the internal circuit switched network, where the local transmission bus is the point of entry and exit. Voice signals transmitted from the port circuit card onto the transmission bus are in digital format. The transmission and coding standard used by all current circuit switched PBX systems is known as Time Division Multiplexing/Pulse Code Modulation (TDM/PCM). To fully understand the workings of the PBX circuit switched network, it is necessary to define the basic terminology (Figure 1).

Figure 1: TDM/PCM.

Multiplexing is the sharing of a common transmission line (bus) for transport of multiple communications signals. A communications transmission bus is a collection of transmission lines used to transport communications signals between endpoints. TDM is a type of multiplexing that combines multiple digital transmission streams by assigning each stream a different time slot in a set of time slots. TDM repeatedly transmits a fixed sequence of time slots over a single transmission bus. In a PBX system, the transmission bus is usually referred to as the TDM bus.

A PBX TDM bus is used to transport digitized voice signals that originate as continuous (analog format) sinusoidal waveform signals. Digital sampling of a continuous audio signal is a technique used to represent the analog waveform in digital bit format. The sampling technique that has become the accepted standard for circuit switched communications is PCM

The Fundamental Enterprise Communications Systems

Current voice communications systems comprise five fundamental product categories:

Key Telephone System (KTS)

Private Branch Exchange (PBX)

Automatic Call Distributor (ACD)

Voice Messaging System (VMS)

Interactive Voice Response (IVR)

The first three categories support call processing and switching functions to enable telephone calls between two or more station users. The latter two product categories are designed to work in conjunction with one of the three core communications switching systems to provide optional services beyond a basic station-to-station call. It’s possible to integrate voice messaging or IVR capabilities into a KTS, PBX, or ACD product design, but most companies don’t. Instead, customers have traditionally chosen to install external, stand-alone systems for messaging and voice response applications. (Many small KTS products have an integrated voice messaging function, but the messaging features are typically not as robust as stand-alone offerings.)

KTS
A KTS is a customer-premises communications system designed to support basic voice applications and relatively small station user requirements. KTS got its name from its historical use with proprietary telephones, known as key telephones, which interface with a central control cabinet known as a Key Service Unit (KSU). The KSU is equipped with the system’s call processing control, port interface cards, and a variety of system/service circuit cards, such as Dual Tone Multi Frequency (DTMF) receivers and Input/Output (I/O) interfaces. The KSU performs central office line connection, intercom functions, paging, and station connections. Its common control elements include a call processor and system memory databases, and its most important function is the provisioning of dial tone. Other basic call processing functions are call answering, dialing, and transaction features such as hold, transfer, call forward, and conference.

Oddly, not all KTS products require a KSU. Instead, the intelligence needed to perform call processing and switching can be built into the circuitry of each key telephone instrument. Such systems are easy to install and maintain, but usually have limited feature and function capabilities, and are acceptable only for customers with modest port-size requirements. The KSU-less system is usually installed to work behind Centrex services offered by local telephone operating companies, which provide the more advanced features and functions through their central office communications switching system.

Common to KTS telephone instruments are designated, programmable key buttons for making and receiving off-premises calls over telephone company line circuits (trunks). The term KTS takes its name from the telephone instrument keys (analogous to typewriter keys) integrated into the product design. The line keys for each telephone instrument have direct access to off-premises telephones. Because trunks are distributed over and shared across groups of select telephones, each station user’s desktop instrument must be provisioned with multiple-line access keys. This is the distinguishing characteristic of a KTS: station user selection and access to designated telephone company lines. A pure KTS product supports only multi-line key telephone instruments and is incapable of supporting industry standard 2500-type single-line analog telephones unless a designated line circuit is dedicated to a single analog telephone instrument. Needless to say, it’s unusual for a customer to configure a KTS in this manner. Small system customers who want station users with 2500-type telephones to have shared access to a pool of trunks do have an alternative: a KTS/Hybrid product (see below), designed to support a mix of proprietary multiple-line button phones and nonproprietary single-line analog phones.

PBX
PBXs aren’t just big KTSs with more features and functions. The two share architecture elements, but PBX systems are designed for more robust functionality, greater growth capacities for ports, traffic, and call processing, and more levels of redundancy. PBX basic architecture includes the common control carrier/cabinet, port carriers/cabinets, and port circuit cards. Peripheral equipment support includes proprietary telephones, both single and multiple line, and industry standard single-line analog telephones.

First, let’s look at the major design and operational difference between a KTS and PBX. PBX stations can place calls over telephone company trunks only with the shared pool access method. The sole exceptions are telephones equipped with an optional private line for inbound and outbound calls. [Note: for reasons too complex to explain, the term line, when used in relation to a PBX system, usually refers to all customer premises equipment peripheral endpoints and is not a reference to a trunk circuit, as it would be with a KTS. The terms station and line are both used for PBX endpoints (telephones, modems, fax terminals) that are not off-premises trunk circuits.]

Hybrid System
Hybrid communications systems also share operating functions common to standard KTS and PBX systems. Original Hybrid systems were designed to more closely resemble KTS rather than PBX systems, although differences between the product categories are diminishing. Like KTS, Hybrid systems are based on a control cabinet similar to a KSU and can support a variety of port circuit boards for interfacing to station and trunk circuit equipment. All Hybrids support multiple-line proprietary key telephones and industry-standard single-line analog telephones. In a Hybrid system, phone access to line circuits is identical to that of a pure KTS, but single-line analog phones access a defined pool (group) of telephone company line circuits. This latter design capability is what distinguishes a pure KTS from a Hybrid system. The Hybrid’s port-oriented architecture design permits custom configurations to suit specific business applications. The architecture and technology design foundations of current Hybrid systems are more similar to PBXs than to KTSs. In fact, features and functions are sometimes difficult to distinguish from more expensive PBX systems.

Unfortunately confusion reigns when it comes to product category typing a communications system as KTS, Hybrid, or PBX. Some manufacturers call their Hybrid offering a KTS/Hybrid and others may refer to it as a Hybrid/PBX. The naming issue gets interesting in the United States because the Federal Communications Commission classifies customer premises communications systems as either KTS or Hybrid based on how single-line telephones access the central office. If the phone can access only one line as programmed by the system administrator, the system is a KTS; if it can access a pooled group of lines, the system is considered a Hybrid. Some manufacturers may even register a single system as both because the call processing software allows configuration flexibility for either pooled- or single-line access from a single-line telephone. Note that designating the product as a KTS, Hybrid, or PBX system may have financial consequences based on telephone company jurisdiction because trunk tariffs for linking a customer premises communications system to the central office can differ between KTS and PBX. (This was more common 15 years ago than it is today.) Ultimately it is the local telephone company that defines the type of system the customer is seeking to connect to the central office.

ACD Systems
The central component of a customer call center is an ACD. ACD systems were originally developed to handle large volumes of incoming calls and automatically route them to designated answering positions. ACD systems are designed and customer programmed to satisfy higher quality of service standards than PBX systems for the following call processing functions:

Screening

Routing

Queuing

Answering

Most PBX systems can be programmed to function as ACD systems, but few ACDs can be programmed to function as PBXs and continue to support most of the latter product’s standard or optional features and functions. Nevertheless, an ACD system shares many of the architecture and feature capabilities of a PBX system. You can think of it as a PBX designed for a very specific application—to distribute incoming calls equitably to a group or groups of answering stations. We usually call ACD answering stations agents, and this is the fundamental difference between PBX and ACD system service: calls handled by a PBX are routed to a specific station user, whereas ACD calls are routed to a group of stations, although call analysis programs can be used to route the call to a specific agent in a group.

ACDs exhibit several architecture design and feature standards that are often not adhered to for PBXs. A true ACD system is based on a nonblocking switch network design, has sufficient call processing power to handle a large volume of complex call types, and has software program- ming features that can screen, route, and distribute calls to agent positions fast and efficiently. Other distinguishing standard ACD system attributes are the ability to support specialized stations, known as supervisor positions, and an integrated MIS reporting system used to monitor, track, and analyze call center operations. All of these features are standard in stand-alone ACD systems, but PBX systems may fail to meet the same standards. Indeed most PBXs must be traffic engineered to support non-blocking switch network access and post-equipped with optional software and external applications servers to all but the most basic ACD feature and MIS capabilities.

Early ACD systems were stand-alone products designed exclusively for a call center environment with large call volumes, such as an airline reservation center. During the early 1980s several PBX systems were designed with optional software that could support basic ACD functions but could not match their performance level. By the 1990s a growing number of PBX systems enhanced with optional ACD software and external MIS reporting systems started to look functionally competitive with stand-alone ACDs. Similar PBX systems with optional ACD packages now control more than 80 percent of the market for call center communications systems. Although stand-alone ACDs hold their own at the high end of the market for large, complex feature and function requirements, PBX and ACD systems totally dominate the call center market segment for systems with fewer than 100 agents. Many KTS/Hybrid systems can also be configured with optional ACD capabilities and are gradually penetrating the very small call center market segment for systems with fewer than 20 agents. Why has the average size of an ACD call center system continually declined over the past 20 years? Customers are realizing that programmable call routing, distribution, and reporting features are beneficial for a variety of nontraditional ACD applications, such as internal help desks and groups of attendant positions. Today’s average ACD installation has only about 60 agent positions, and that number is declining.

Voice Messaging System
Most enterprise environments use a VMS as the primary call coverage point for unanswered calls, but you’d be selling the VMS short to think of it only in terms of message record and store. A good VMS mailbox can:

Be programmed for different greetings

Offer incoming callers a menu of options for leaving a message or transferring to another station

Act as an automated attendant position to answer and route calls

Serve as an automated information system or an outbound call messaging system

VMSs can be interfaced to almost all current generation KTS/Hybrid, PBX, and ACD systems by using a signaling link between the VMS, the switching system, and its voice communications channels. The signaling link is usually referred to as the voice mail system interface or, in standards parlance, the Station Message Detail Interface (SMDI). Among other functions, the link activates the message-waiting indication at a user station. Its voice communications channels will be based on 2500-type analog station interfaces, a standard interface supported by all systems.

Here’s what happens when a caller leaves a message. VMS coders/decoders (codecs) take the transmitted voice communications signals from the switching system, digitize them, and compress them for storage purposes. The same codecs will convert the digitally compressed messages back to analog format for station user playback. The voice quality of VMS playback largely depends on compression algorithms that may degrade the original message to optimize storage capabilities by using a low sampling rate or bit scheme. Filters and automatic gain control improve sound quality, but if stored messages are forwarded to another station or broadcast to hundreds of stations, further digital compression occurs as messages pass through interim mailboxes. Moreover, when VMS is used behind IP telephony, playback quality weakens if stored message transmission is packetized using IP codecs. This is a technical issue being addressed by designers and developers of the new IP-PBX systems.

Because the market trend is toward Local Area Network (LAN)–connected application servers, the traditional method for setting up signaling between the switch and the external VMS is slowly changing. The new, improved method is to insert an Ethernet TCP/IP link between the switching system and the LAN-connected VMS. At the same time, the newer VMS server design is also driving the market for Unified Messaging systems (UMSs). Many people think that first-generation UMS failed to take hold because they were based largely on traditional VM systems design and user operation. In contrast, second-generation systems are based largely on e-mail server design and user operation and are gaining greater market acceptance. Several new client/server PBX system designs include the VMS or UMS function as an integrated feature capability.

Interactive Voice Response System
An IVR system is a communications system product that typically functions as an intermediary between a PBX system and external computer databases. An IVR can also function as a stand-alone enterprise communications system, without a PBX system, because it can support standard PSTN trunk interfaces. Analog and digital trunk interfaces are supported by most IVR systems to connect to the customer premises PBX system or directly to the PSTN. IVR systems are used for several customer applications, including automated voice response and feedback, automated directory, call routing.

The primary system link for an IVR system is not the switching system, but an external computer system. The IVR mediates between voice callers and computer databases with customer-written scripts and menus prompting callers to respond to prompts with dialpad entries on their phones. The IVR system interprets the DTMF signals from the dialpad and is programmed to respond with another voice prompt, a recorded announcement, or a “spoken” answer to the caller’s inquiry. IVR speech is based on a text-to-speech programming algorithm, for which appropriate “answers” can be stored in the IVR database (which usually has limited memory storage capacity) or (more commonly) an external computer database.

Some IVR systems with ASR capabilities don’t require DTMF input to respond to caller voice commands and questions. Voice-based interaction with the IVR can significantly speed up the IVR transaction process by bypassing many call prompt levels of the programming tree. There are also many instances when callers find it easier and faster to speak digits or words instead of using the dialpad to enter digits in response to a call prompt command. ASR systems have been famously slow to penetrate the market because of reliability and cost barriers, but recent advances in programming and the declining cost of digital signal processors have made ASR more prevalent. We anticipate that the next major technology advance for automated response systems is speaker verification, an important capability to simplify the system interaction process and raise security levels.

Convergence of KTS/Hybrid and PBX Systems
KTS and PBX architectures began their gradual convergence of design and function with the introduction of the first KTS/Hybrid systems. The early 1980s saw major differences in call processing, switching, and port cabinet-design pure KTSs and PBXs. KTSs used analog switching and transmission standards and were based on a common control system design, with limited traffic, processing, and port capacities. Features were necessarily limited, and options such as multiple system networking and ACD were unheard of. Migration between KTS models—even on the same manufacturer’s product platform—usually required KSU fork-lifts. However, most PBX systems of the day employed digital switching and transmission standards, with digital desktop telephone and trunk interface support. PBXs were designed to handle greater traffic, processing, and port expansion requirements and offered numerous advanced features not available on any KTS.

Hybrid systems began to blur these category differences as PBX design technology and feature capabilities were applied to a KTS-like platform, nudging Hybrid switching network design away from the traditional dedicated line access and intercom path. The result more closely resembled the digital Time Division Multiplexing (TDM) bus design used by PBX systems. Hybrid systems could support customer port capacities far in excess of KTSs, some basic networking options, and call center applications.

By the mid-1990s many customers couldn’t tell the difference between a Hybrid system and a full-function PBX system because the design platform and feature sets were so similar as to be indistinguishable for all but the most unique customer requirements. High-end Hybrid systems could support customer port requirements up to several user stations. They accommodated networking options such as digital T1-carrier trunk interfaces and ISDN PRI services, ACD options including MIS reporting packages similar to the early PBX/ACD systems, and some integrated voice messaging and wireless communications options. Hybrid system digital telephones were more advanced, often offering customers a greater selection of multiple-line key and display models. It was not unusual for Hybrid telephones to offer the large display fields and soft-key feature buttons that are only now becoming a PBX standard.

Today’s Hybrids are often larger in capacity than many entry-level PBX systems and can support most, if not all, of the same features and functions for the majority of customer requirements. They can be intelligently networked, and they can satisfy complex call center requirements. Some manufacturers offer the same telephone models for both their Hybrid and PBX models, allowing customers a migration path between the two platforms, and most manufacturers have announced that their recently shipped IP telephones will work behind either systems.

As Hybrid systems expanded, PBX systems grew smaller. Until recently PBX systems were not priced to be competitive against Hybrid systems at the same station sizes. (The target market for competitively priced PBX systems usually began at 40 stations.) This PBX price problem was created by the higher cost of its more robust common control cabinet or carrier. During the past few years the PBX manufacturers have responded by redesigning their common control systems and downsizing the control cabinet/carrier hardware equipment. Although today’s PBX systems remain higher priced than a KTS/Hybrid configured for the same number of stations and trunks, the price differential has been continually shrinking. There is currently a 10 to 15 percent price difference between the two system platforms compared with 25 to 50 percent a decade ago. Today there are entry PBX models designed around a single printed circuit board for call processing, memory, and switch network functions. These new models are price competitive but port limited, usually designed for customers with 20 to 40 station size requirements at initial installation, but equipped with the call processing capability to support all of the features and functions of PBX models many times their port capacity.

PBX systems based on client/server designs are targeted primarily at the small/medium enterprise (SME) market, defined as ranging from about 20 stations to about 120 station, and overlaps with the Hybrid system market. A typical Hybrid system installation is about 25 stations. Shipments of PBX systems based on client/server designs are currently averaging about the same station size as Hybrid systems, and few have been installed at customer locations with port requirements above 100 stations. As you might expect, most of the systems being replaced by the client/server PBX systems are Hybrid systems, against which the newer PBXs can offer more bundled applications and support (either current or planned) for IP endpoints. The current generation of installed-base Hybrid systems may someday support IP endpoints but only after costly system upgrades. For the vast majority of users, it will be less expensive to install a new system than to upgrade the old. The first generation of IP telephony systems designed for the very small KTS/Hybrid customer is just beginning to make its way into the market.

Under the circumstances, we can predict that the traditional Hybrid system eventually will be replaced by the new communications system, client/server design, competitive product offerings, or upgrades to the system manufacturer’s new design platform. But this picture is also affected by customer size. Almost all market demand forecasts predict that the new PBX designs will be far more successful in smaller rather than larger line size markets, because the latter customer segment is less likely to replace a reliable and upgradeable installed system for a system that has not yet proved as reliable. The cost factors for large system replacement go far beyond the system purchase and installation price. Too much is at stake for large line size customers to risk the lesser reliability level of an OEM server running an operating system, such as Windows NT, not originally developed for real-time industrial applications. Only a few of the new client/server systems are built on more reliable and flexible operating systems, such VX Works, and custom-designed common control elements. Branch offices may sacrifice reliability for a less expensive system, but a large line size corporate or regional headquarters customer would not.

Fundamentals of PBX Circuit Switching

Time Division Multiplexing
The core design element of a traditional digital PBX is the local transmission bus that connects to a port circuit card. Many port circuit cards may share a common local transmission bus, and a PBX system may have many local buses dedicated to designated port circuit cards housed in different port carrier shelves and/or cabinets. Port circuit cards are used to connect peripheral equipment devices, such as telephones and telephone company trunk circuits, to the internal circuit switched network, where the local transmission bus is the point of entry and exit. Voice signals transmitted from the port circuit card onto the transmission bus are in digital format. The transmission and coding standard used by all current circuit switched PBX systems is known as Time Division Multiplexing/Pulse Code Modulation (TDM/PCM). To fully understand the workings of the PBX circuit switched network, it is necessary to define the basic terminology (Figure 1).


Figure 1: TDM/PCM.

Multiplexing is the sharing of a common transmission line (bus) for transport of multiple communications signals. A communications transmission bus is a collection of transmission lines used to transport communications signals between endpoints. TDM is a type of multiplexing that combines multiple digital transmission streams by assigning each stream a different time slot in a set of time slots. TDM repeatedly transmits a fixed sequence of time slots over a single transmission bus. In a PBX system, the transmission bus is usually referred to as the TDM bus.

A PBX TDM bus is used to transport digitized voice signals that originate as continuous (analog format) sinusoidal waveform signals. Digital sampling of a continuous audio signal is a technique used to represent the analog waveform in digital bit format. The sampling technique that has become the accepted standard for circuit switched communications is PCM.

Pulse Code Modulation
PCM is a sampling technique for digitizing the analog voice-originated audio signals. PCM samples the original analog signal 8,000 times a second. This is more commonly referred to as 8-KHz sampling. The sampling rate used to code voice audio signals is based on the frequency range of the original signal. To accurately represent an analog signal in digital format, it is necessary to use a sampling rate twice the maximum analog signal frequency, a calculation based on the Shannon theorem. The maximum frequency of human voice is about 3.1 KHz. This frequency was rounded up to 4 KHz for ease of engineering design, resulting in an 8-KHz (2 × 4 KHz) sampling rate for digitizing voice audio signals. An 8-KHz sampling rate translates into a one sample every 125 microseconds (8 KHz–1; Figure 2).


Figure 2: PCM encoding.

Each digital sample is represented by an 8-bit word (28 = 256 sample levels) that measures the amplitude of the signal. The amplitude of the signal is based on the power (expressed in units of voltage) of the electri- cal signal generated by the telephone transmitter/receiver in the handset. This signaling technique has become known as Digital Signaling 0 (DS0), or 64-Kbps (8 bits × 8 KHz) channel transmission format. The term DS0 was defined based on the Digital Signaling 1 (DS1) format used to describe a digital T1-carrier communications circuit supporting 24 64-Kbps communications channels.

The PCM samples generated from each communications system port are transmitted onto the TDM bus in a continuously rotating sequence based on the time slot assignments given to each port circuit interface (see below). Only a single PCM word sample is transmitted at a time; that is the entire electrical transmission line is reserved for use by only one port circuit for transmission of its sample signal. The PBX processing system monitors each port circuit’s transmission time assignment in the rotating sequence, controls when the sample is transmitted, and coordinates transmission of the sample between the originating and destination endpoints.

There are two standards for coding the signal sample level. The Mu-Law standard is used in North America and Japan, and the A-Law standard is used in most other countries throughout the world, although each uses the 64-Kbps transmission format. For this reason, PBX systems must be designed and programmed for different geographic markets. Using firmware downloads, system vendors and customers can program their PBX systems to support the local PCM standard. The early digital PBX systems used different hardware equipment based on the location of the installation.

To summarize the fundamentals of PCM:

1. 4-KHz analog voice signals are sampled 8,000 times per second (8-KHz sampling rate)

2. Each sample produces an 8-bit word number (e.g., 11100010)

3. 8-bit samples are transmitted onto the TDM bus at a 64-Kbps transmission rate

4. The samples from each port circuit are transmitted in a continuously rotating sequence