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

Messaging | Feature/Function Enhancements

VMSs were introduced to the market in the early 1980s, and originally worked as stand-alone systems behind PBX systems. Although the first VMSs were designed and marketed by third-party suppliers, several of the leading PBX manufacturers eventually entered the market with products of their own design. Rolm and AT&T were among the first PBX manufacturers to enter the voice messaging market with products designed to work behind their own communications systems, although they could also be engineered as stand-alone systems to work behind other suppliers’ PBX systems.

Northern Telecom, one of the leading PBX suppliers, came late to the VMS market during the late 1980s, but when it introduced Meridian Mail it became the first messaging system to be fully integrated within the PBX system design. Meridian Mail used the Meridian 1 processing and switching network backplane for supporting PBX station user messaging applications. The Meridian Mail Module was installed as another cabinet stack in the Meridian 1 and tightly integrated within the overall PBX system design. Instead of using analog station interfaces and a dedicated data signaling link between the PBX system and adjunct voice messaging cabinet, Meridian Mail ports appeared to the Meridian 1 switching network as just another station port, and signaling between the Meridian Mail Module and the Meridian 1 common control complex was transmitted over the internal system processor bus. AT&T followed Northern Telecom’s example and later redesigned its Audix VMS as a multiple card slot equipment module to be installed within its Definity PBX system. The Definity Audix option offered most of the features and functions available on the larger, stand-alone Audix (later Intuity Audix) system at a reduced price.

During the early 1990s VMSs were redesigned to support integrated messaging applications with e-mail servers. The concept of a UMS designed to support voice and e-mail messaging, with both message mediums sharing a common directory and storage system, was also introduced in the early 1990s. Although demand for the enhanced messaging system designs has been limited to date, there are many productivity and cost benefits attributable to using one mailbox for all types of messages and having a single interface to the mailbox from either a telephone or PC client.

Recognizing the competitive advantage of bundling messaging applications within the PBX system, several recent start-up companies with PBX client/server designs, such as Altigen and NBX (since acquired by 3Com), integrated a UMS application running off the main system server that also provided basic PBX communications features and functions. Recently, Avaya integrated the capabilities of a full-function Intuity Audix system into the main call processing board of its small Definity One PBX system and included the Intuity Integrated Messaging appli- cation on the same board. The Altigen, 3Com, and Avaya PBX systems with the bundled messaging capabilities are designed for small/intermediate customer line size requirements, and the message storage capacities and access ports are limited. PBX systems designed for large and very large customer port requirements would not be able to integrate the messaging application into the main common control complex without affecting the basic communications responsibilities of the system. Dedicated messaging application servers will likely be the optimal solution for higher-end PBX customers, even when IP-PBX client/server system designs become standard by the end of this decade.

Figure 1: Avaya speech access with unified messenger architecture.