Telephone Network Interfacing


3.10

INTRODUCTION

From earnest political talk presentations to raucous morning shows, listener involvement via telephone is an important programming element at many radio and television stations. When we want to create a two-way connection with our listeners, we will probably be using the dial-up telephone network.

Radio news departments rely extensively upon phoners to get reporters and newsmakers on the air in a timely fashion. Why are the people who run local TV news so concerned with avoiding the dreaded talking head—that is, the anchor simply reading a story into the camera? Because they’ve discovered that being there is better. The same is true for radio.

Today, integrated services digital network (ISDN) lines combined with modern audio compression techniques permit instant full fidelity remotes from almost anywhere in the world.

This chapter will explore all of the ways to integrate the ubiquitous telephone network into broadcast operations. First, we’ll learn about the nature of the various services available from telephone companies. Then we’ll investigate ways to interface them to our station facilities.

 

THE TELEPHONE NETWORK

As we transition our broadcast facilities to digital systems, it is interesting to note that the standard voice telephone network is almost entirely digitized and has been so for many years. The watershed event was Illinois Bell’s 1962 installation of a T-carrier system— the first widespread commercial application of digital audio. Telephone engineers appreciate digital technology for the same reason broadcasters do: reduced susceptibility to noise and other disturbances, and improved ability to switch, monitor and maintain the circuits.

While the worldwide dial-up telephone network is an amazing achievement, it is mostly made from a simple ubiquitous element: digital circuit-switched channels of 64 kbps each.Circuit-switched means that the channel is connected end-to-end with the entire capacity available for the duration of the call. (This is in contrast to packet-switched systems, such as the Internet, where capacity is shared among users and there is usually no guaranteed bandwidth.)

While most of the network infrastructure is digital, the last-mile copper connections from the central office to the customer site mostly are not. The vast majority of users interface to the network via an analog technology that is little different from that employed in Alexander Bell’s days. This is beginning to change with the introduction of digital last-mile technologies like ISDN, T-1, and an Asynchronous Digital Subscriber Line (ADSL).

Incidentally, in industry jargon, your local phone company is a local exchange carrier (LEC) or simply a telco. A long distance company is an inter-exchange carrier (IEC).

Speech Coding

The bit rate of 64 kbps was chosen to support phonegrade speech audio encoded using a modified pulse code modulation (PCM) technique. When we make a plain old telephone (POTS) call, our speech is sampled at an 8 kHz rate and encoded into a digital word 8 bits long. Telco engineers call this 64 kpbs bitstream a digital signal level 0 (DS-0) channel.

Theword lengthis whatdetermines dynamicrange— and 8 bits would only permit 48 dB were it used in standard PCM linear fashion. A primitive kind of compression is used to stretch the dynamic range:mLaw in North America and much of Asia, and A-law in Europe (see Figure 3.10-1). This is a scheme that equalizes the step size in dB terms across the dynamic range—a smaller step size on low level signals reduces quantization noise and improves effective dynamic range to the equivalent ofabout13bits.Thus,thequantizationnoise(anddistortion) is approximately a fixed percentage of the signal amplitude, regardless of its level.

The process of conversion and companding is done in specialized analog-to-digital (A/D) and digital-to-analog (D/A) integrated circuits called codecs(CODer/ DECoders). The method is specified by the International Telecommunications Union (ITU) as standard G.711.

 

 pcm-coding    4-wire-signals

 

2-Wire and 4-Wire

Both speech directions are mixed together on the usual analog lines with which we are most familiar, but this is not the way signals are handled within the telephone transmission and switching network. Noncopper transmission media such as microwave radio, satellite and fiber-optic cables are one-way only, so the paths must be kept independent. Even when copper is used, long-distance links are kept separated so that amplification can be inserted. A standard analog POTS circuit is 2-wire, because it arrives on two wires. The network is internally 4-wire, so named because in the past, a 4-wire circuit needed a separate wire pair for each of the send and receive transmission directions— four wires altogether.

The Traditional Analog Line

The traditional telephone lines provided by the phone company are known officially as subscriber loops, trunks or simply CO (central office) lines. (Trunks used to refer only to lines destined for private branch exchange (PBX) systems and may have included special signaling as well.)

Because these are 2-wire circuits, the CO uses a 2-to- 4-wire converter (also called a hybrid) to interface the analog lines to its internal 4-wire system, as shown in Figure 3.10-2. This process happens on the line card, which is also responsible for digitization, talk battery insertion, off-hook detection, and ring generation.

Talk Battery and Ringing

The talk battery direct current (dc) voltage and the conversation audio appear together on the phone pair. The talk battery leaves the exchange at 148 V and is limited to 20–50 mA by a series resistor. The resistor’s value is selected to complement the resistance of the loop. The dc resistance of the loop itself varies from a few to 1,300 V depending on length. Because of this series resistance, when a line is off-hook, its voltage at the customer equipment drops to around 112 V.

For ringing, an ac voltage of 90 vrms at 20 Hz is superimposed on the line. Talk battery is maintained during ringing, so that the resulting signal has a sinusoidal shape shifted 48 V to the negative.

Talk signals are ac coupled with nominal impedance of 600V. However, some CO equipment uses complex impedancecoupling,andthenatureofthetelephonenetwork usually results in the actual impedance as presented to the user rarely being the specified simple 600V. This turns out to be an important issue for broadcast interfacing, which we will discuss in detail later. The basic parameters are summarized in Table 3.10-1.

Frequency Response

For ordinary subscriber loops, the phone company specifies a frequency response of 300 Hz to 3.4 kHz. In the not-too-distant past when all local calls were connected at the exchange by metallic contacts, better frequency response was likely to be had on many conversations. Today almost all calls are digitized and are strictly limited to a 3.4 kHz bandwidth by the sharp low-pass filters required for proper digitization. The phone network’s 8 kHz sampling rate permits a theoretical Nyquist frequency of 4 kHz, but a 600 Hz transition band is necessary for anti-aliasing and reconstruction filtering (see Figure 3.10-3).

phone-loop-charistics

Noise and Level

A 1971 Bell System survey of the phone network nationwide determined that the average conversation had a level of 116 dBm. Of course, as anyone who has wrestled with broadcast-to-telco interfacing knows, incoming level varies tremendously, with a range of perhaps 140 to 14 dBm, as illustrated in Figure 3.10-4.

Send audio (that is, audio fed into the telephone line) must be limited to 19 dBm as specified in Part 68 of the Federal Communication Commission (FCC) Rules. Audio loss on any given local loop is limited by tariff to 8 dB or less. This loss limit, however, applies only to the loop from the CO to the subscriber and does not include the rest of the signal path. Also, the 8 dB loss may occur at each end of a conversation path: once at the calling party end and again at the called party end, for a total loss of 16 dB.

The phone engineering people measure noise upside-down, defining a reference noise floor and then measuring up from there. The reference noise level is one picowatt, which corresponds to 190 dBm. Thus, a noise level of 160 dB relative to 0 dBm would be reported as 30 dBrn noise (dBrn4dB above reference noise). Note that, according to this method, the higher this number, the worse the noise.

Be aware also that when telephone people measure noise, they are measuring only idle channel noise.This is an important difference, since in digital systems idle channel noise is not the same as the traditional (S/N) measurement in analog systems. Noise in a digital system will generally increase when a signal is present. This effect is called modulation or quantization noise and is primarily dependent upon the number of bits used for quantization.

A C-message weight filter is employed when measuring phone line signal-to-noise ratio (S/N). (See Figure 3.10-5.) The C-message curve was developed years ago to simulate the frequency response of an old-style telephone earpiece and, accordingly, it has considerable low-frequency roll-off. This means that a line can have significant hum and other low frequency noise and can still meet the officially mandated noise specs. While this makes life easier for the phone company technicians, it can be troublesome when a broadcaster is trying to use phone audio on the air. If noise is a serious problem, try to get the technician to switch the noise meter to the flat position. The measuring set usually does have this option available.

low-pass-graph

signal-noise-reference

DTMF Tone Dialing

Dual Tone Multiple Frequency (DTMF) dialing uses two frequencies for each digit in order to avoid talkoff—that is, the tone detector accidentally sensing voice as a dial command. In addition, the frequencies were carefully chosen to avoid problems with harmonic distortion causing false detection. There are four low group frequencies, one for each button row, and four high group frequencies, with one assigned to each column as shown in Figure 3.10-6. Tolerance is61.5% for the encoder and 62% for the digit receiver. The time required to recognize any digit tone is 50 msec with an interdigit interval of another 50 ms. Low group tones are supposed to be sent at a level between 110 and 16 dBm; ideally, tones in the high group are transmitted with 2 dB greater level in order to compensate for high-frequency roll-off in the phone line.

c-message

dtmf-tone

Loop Start and Ground Start

Central office lines come in two basic configurations: loop start and ground start. Loop start is the kind that is most common. In this kind of circuit, the CO provides talk battery to the line at all times and detects that an off-hook condition is occurring when the terminal equipment connects and causes current to flow between the tip and ring. (Incidentally, the terms tip and ring originated with the description of the circuits being on the tip and ring of the patch cords that used to be used by telephone operators.) With ground start circuits, the CO waits for a connection from the ring wire to ground before connecting talk battery, at which time the terminal equipment removes the ground connection to establish a balanced talk path. When the calling party hangs up, a ground start circuit removes talk battery. A loop start circuit may or may not provide a momentary interruption or reversal of the talk battery when the calling party terminates.

Many PBXs are designed to work with the ground start circuits because the possibility of collision is reduced. Collision occurs when the phone system tries to seize a line for an outgoing call just as that line is ringing in.

Disconnection: Calling Party Control

Loop-current interruption occurs on most telco lines when the calling party hangs up. It is sometimes referred to as calling party control (CPC), since the calling party controls your equipment when he hangs up. The CPC may turn off an answering machine, for example, or extinguish the winking light on a held line on a key phone. The CPC interruption was probably never intentional, having been a by-product of early mechanically switched relay-controlled exchanges. Thus, some phone lines do not provide this function or they provide it unreliably. However, with the proliferation of answering machines that rely upon CPC, most central office equipment now has this capability designed in. In some cases, it is necessary to specifically request this feature from the phone company on a per line basis.

Loop-current reversal, on the other hand, has long been a phone company signaling method. First used between the telco’s own central offices, loop-reversal was later employed to communicate with some large premises PBX systems. Thus, lines that are set up for PBX use, or originate at central offices with large concentrations of business customers, sometimes use this method. (However, the preferred and more modern situation for PBX control is to use ground-start lines.)

While most exchanges do provide CPC, there are some that do not reliably provide it or provide it after a variable time delay. Most PBXs do not generate it. However, every telco CO in the United States eventually returns dial tone to its lines when the calling party hangs up. Thus, we can use the presence of dial tone as a back up to cause a disconnect when the loopcurrent detection methods fail. An important consideration is to prevent false talk-off from noise, applause or other spectrally rich audio. Using software based statistical methods ensures that the dial tone is really present before terminating the connection.

Caller ID

Caller ID (CID) allows you to know the phone number of the caller. This capability is useful for call-in shows, where it might be desirable to deny access to problem callers. The technology is simple. Between the first and second ring, the information is sent in a packet using a 1200-baud modem. This is exactly the same modulation scheme used in normal computer modems operating at this rate. Customer equipment normally suppresses the first ring so that the answering user does not take the call before the CID information is fully transmitted.

Loading Coils

A typical #24 gauge phone pair attenuates a 3 kHz signal 2.5 dB per mile due to capacitive effects. On an 8 mile (12.9 km) long line, high-frequency attenuation would thus be 20 dB, a significant amplitude distortion. Loading coils are toroidal inductors, which counter the effects of the phone pair’s natural capacitance. While the coils are effective at flattening out the response within the voice band, the roll-off above 3.5 kHz is devastating, as shown in Figure 3.10-7.

Physically, load coil banks are long cylinders, with the individual donut-like coils stacked one on top of the other inside. They are typically placed at 3,000 (.9 km), 4,500 (1.4 km), or 6,000 (1.8 km) ft intervals along the phone cables. Generally, loading coils are found only on cables of greater than 3 miles (4.8 km) in length.

As we shall see, loading coils can create problems for the hybrids used in broadcast interfaces.

4-Wire Circuits

It is possible to purchase analog 4-wire circuits from telcos. These are used where it is desirable to maintain separation in the two speech paths. They are not dialup, but rather end-to-end hardwired. This service has traditionally been used by television remote trucks for connection of remote production intercom systems. With the introduction of digital hybrid interfaces, use of this approach has been in decline. ISDN offers 4-wire capability at a lower cost and with fewer hassles, so it will probably supplant these analog lines over time.

loading-coil-graph

Foreign Exchange (FX) Loops

FX provides local telephone service from a central office that is outside (foreign to) the subscriber’s exchange area. If a station is located in the suburbs and the choke network central office is downtown, FX loops will be needed to connect your lines. When the phone is picked up, you get dial tone not from your local suburban CO, but from the downtown office. FX service is also sometimes used to extend your coverage into another city, so that people can call the station without paying a toll charge and calls can be made within that city without incurring toll charges. For instance, if the studio is in Cleveland and the goal is to serve listeners in Akron as if they were local, FX service could be the answer.

An FX loop is a 4-wire circuit with hybrids at each end, at each terminating central office. Since FX loops add an extra layer of hardware to the phone audio, they are another source of problems for on-air interfacing. They usually are engineered to have a few dB loss and they add to the impedance complexity of the line.

FX circuits are usually expensive and pose certain technical challenges. Since, as we will learn later in this chapter, hybrids are imperfect, a potential for a special kind of feedback called singing exists. This results from the inevitable leakage from the send to the receive ports at each hybrid. The phone people solve this problem by inserting a pad—anywhere from 5–8 dB is common.

Choke Networks

Most stations need special high volume exchanges for their contest and request lines. This requirement probably results from the days when aggressive program directors (PDs) desired the publicity that burning out a phone exchange would generate.

The choke network works by diverting calls beginning with the unique choke prefix around the local serving central office and sending them directly to the choke switching exchange, usually located downtown (see Figure 3.10-8). The phone company dedicates very few talk paths (wire trunks or special carrier equipment) to the task of connecting the caller’s serving CO choke ports to the choke exchange. The usual switching and routing process is bypassed. Unfortunately, only a very limited number of paths are generally provided. In the densely populated Los Angeles area, for instance, only three connections exist from most central offices. In addition, the poorest facilities are often given over to the high volume service.

Generally, unless you are near the choke central office, the FX circuits previously described are employed to connect the choke CO to your serving CO. This is one of the reasons why choke circuits often have a lower level than standard lines. Because of their higher complexity, choke lines also usually have bumpier impedance curves, making good hybrid performance difficult to achieve due to the problem of finding appropriate balancing network values. This is especially a problem with simple analog hybrids.

In some areas, FX circuits are being replaced by internal call forwarding. This means that a published number is actually being software forwarded to a real number originating from your local serving CO. The main advantage to this approach is lower cost, since you do not have to pay the premium for the FX circuit. However, there usually is a smaller call-forwarding charge.

choke-network

ISDN: Basic Rate Interface (BRI)

ISDN allows a direct digital connection to the telephone network. In addition to the quality advantages digital transmission offers for basic voice service, users may bypass the normal POTS speech coding methods and supply their own much better algorithms, such as those standardized by Moving Pictures Expert Group (MPEG). MPEG is an organization involved in standardizing audio coding. Another characteristic of ISDN important to broadcasters is that the B channels are true full-duplex, with absolutely no cross-connection between the send and receive signal paths.

ISDN is now widely available and is growing in popularity—mostly because of its value for high-speed Internet connectivity. Web surfers may implement direct digital links without the bottleneck caused by inef- ficient, slow modems. An ISDN BRI has 128 kbps raw capacity. Compare this to the speed possible with a 33.6 kbps modem and it becomes evident why the promise of ISDN creates so much excitement among people who need fast access to the net.

With a BRI line, you get two 64 kbps voice or data channels, called “B” or bearer channels, and one 16 kbps “D” or data channel on a single telephone pair (see Figure 3.10-9). The D data channel is the path between the central office and terminal equipment that is used for call set-up and status communication and is usually not available to the user.

The S and U Interfaces

The line from the central office is a single copper pair physically identical to a POTS line. When it arrives at the subscriber, this is called the “U” interface. The U interface converts to an S/T interface with a small box called an “NT-1.” In the United States, NT-1 functionality is usually included in the terminal equipment. In Europe, the telephone company provides the NT-1. Only one NT-1 may be connected to a U interface, but as many as eight terminals may be paralleled onto an S bus.

Professional equipment should usually provide access to the S interface, making it possible to parallel multiple terminals. You can use either an external NT-1, or the equipment may have an internal NT-1 with both U and S/T connectors.

Terminal Adapters

A terminal adapter (TA) is the equipment that interfaces to the ISDN line, providing call set up and protocol conversion functions. A traditional TA has an ISDN connection on one end and one or two bit stream ports on the other, usually using the V.25 or X.21 connectors. Modern broadcast equipment combines this capability with the audio encoding equipment into one integrated unit.

SPIDs

Service profile identification numbers (SPIDs) are only required when you are using the National I-1 ISDN protocol in the United States. This number is given to the user by the phone company and must be entered into the TA in order for the connection to function. SPIDs usually consist of the phone number plus a few prefix or suffix digits.

The intention of the SPID is to allow the telco equipment to automatically adapt to various user requirements by sensing different SPIDs from each type or configuration of user terminal. For instance, multibutton phones could retain function assignments when moving from line to line. In this case, the line number would probably not be used as the SPID. None of this matters with our application, but we must enter the SPIDs nevertheless. (Over time, it may be possible that a standard SPID could be used for all broadcast codec applications. A proposal that would allow this is being considered.)

If you are using the National I-1 protocol, your telco service representative must give you one or two SPID numbers for each line ordered. You will get one SPID for each B channel you need. Upon power-up, connection of the ISDN line or boot, the TA and the telco equipment go through an initialization/identification routine. The TA sends the SPID and, if it is correct, the network signals this fact. Thereafter, the SPID is not sent again to the switch. You must have this SPID number, and it must be 100% correct, or the system will not work. Do not let the installer depart without leaving your SPID number(s).

Directory Numbers (DNs)

Directory numbers (DNs) are the telephone numbers assigned to the ISDN line. You may be assigned one or two, depending upon the line configuration. If you have two active ISDN B channels, you will usually have two DNs. However, the physical channels are independent from the logical numbers. A call coming in on the second number will be assigned the first physical B channel, if it is not already occupied. Therefore, there must be some way for the TA to sort out which call goes to which channel/line. The DN is used for this function.

When a call rings in, it contains set-up information, which includes the DN that was dialed by the originating caller. The last seven digits are matched with the DNs programmed into the TA and the proper assignment is made. However, it is not usually necessary to explicitly enter them, as they are almost always contained within the SPID, and most TAs are smart enough to look there first. The only time a DN must be entered is in the very rare case where the last seven digits of the DN are not included somewhere within the SPID. When DNs are required, only the last seven digits need be entered.

Digital Long-Distance

Long-distance connectivity is routinely available in most parts of the United States from the big-three carriers: AT&T, Sprint and MCI. The “dial 1`” default carrier may be chosen at the time you order the line, just as with traditional voice lines. Also, just as with voice lines, you may usually choose a carrier on a per call basis by prefixing the number with the 1010XXX carrier selection code. You must dial the full number, including the 1 or 011 ` country code following the prefix.

Here is a hot tip: You can save a lot of money by arranging a special plan with your long-distance (LD) carrier. When you use 1` dialing without contacting your LD carrier, you are generally put into a standard rate plan that has the highest cost of any of the pricing tiers.

Some long-distance connections are limited to 56 kbps/channel. This arises from a quirk of the older telephone infrastructure. The channel banks that have been widely employed in the long-distance network have a native 64 kbps capability but rob the low order PCM bit on every sixth frame in order to convey supervision information (on-hook/off-hook and dial pulses). This limitation is becoming more rare as equipment is upgraded, but there is no way to know for sure in advance.

CSD and CSV

Recall that each ISDN BRI has two possible B channels. It is possible to order a line with one or both of the B channels enabled, and each may be enabled for voice and/or data use. Phone terminology for this class of service is circuit switched voice (CSV) and circuit switched data (CSD). (Both are in contrast to packet switched data (PSD) which is possible but irrelevant to this discussion.)

CSV is for standard voice phone service and allows ISDN to interwork with analog phone lines and phones. CSD is required for MPEG codec connections. Even though you may be sending voice, the codec bit stream output looks like computer data to the phone network.

Even for MPEG codec applications, you may want POTS speech capability, since some support this feature. Therefore, you may want to order CSV as well as CSD on one or both B channels. To get a line with one B channel to be used with either hi-fi or speech, you would request an ISDN BRI 1B`D line with CSV/CSD capability. For both B channels, you would order an ISDN BRI 2B`D line with CSV/CSD on both channels; if you do not need voice possibility on the channels, you want 2B`D with only CSD enabled.

Protocols

In a perfect world, all ISDN terminal equipment would work with all ISDN lines, without regard for such arcana as 5ESS, DMS100, CSV/CSD, SPIDs, etc. Unfortunately, the ISDN standard has been evolving for years and has only recently begun to settle down. And, sadly, there will remain different standards for the Unites States and Europe.

The telco network and the TA communicate via a protocol—the language the user equipment and the telephone network use to converse (on the D channel) for setting up calls and the like. This is where you will find differences, since the protocol depends upon the central office equipment and the standards that it follows.

In the United States, telephone companies use either AT&T 5ESS, Northern Telecom DMS100, or Siemens EWSD switches. Each of these can support the National I-1 protocol standard, which has been specified by Bellcore. However, both AT&T and Northern Telecom had versions of ISDN which pre-date the NI-1 standard and some switches have not been upgraded to the new format. There is also a newer NI-2 standard, but it is designed to be compatible with NI-1 for all of the basic functions.

In Europe, the common protocol is Euro-ISDN, following the ETS300 standards. It is an apparently successful attempt at having all of the European telephone networks use a single, compatible protocol. The telco authorities in most countries have adopted it already, with most of the rest planning to do so.

 

T-1 Digital Service

As with ISDN, T-1 is possible because an ordinary copper phone pair can carry a much wider signal than the 3.4 kHz required for a single voice conversation. Indeed, a pure metallic path of reasonable length is easily capable of passing frequencies in excess of 100 kHz. Thus, digitization and multiplexing can be used to carry a number of voice channels over a single pair of wires.

Introduction to T-1

To create the T-1 bit stream, 24 64 kbps DS-0 channels are assembled serially and the equivalent of another 8 kbps channel is added for synchronization (see Figure 3.10-10). Thus the ultimate data rate becomes 1.544 mbps, a rate also called DS-1. The signal is then converted into a digital bipolar bit stream in a special format called binary 8-zeroes suppression (B8ZS). The voltage is modulated between 13 V and `3 V.

Most LD carriers offer service on T-1 connected directly to their point of presence (POP). Because the LD carrier does not have to pay the usual fee to the local telco for routing over their CO and lines, the customer cost can be lower.

bit-stream

Using T-1: The Customer Provided Equipment (CPE)

Despite the difference in capacity and service, T-1 arrives at the end user site as two conventional copper pairs: one for the data send and another for receive. The physical connector used to be a DB-15 type, but the current standard is the common RJ-48C, an 8- position modular plug. Figure 3.10-11 shows both types.

Here are the usual components of a terminal system for a T-1 circuit:

  • The CSU and DSU. The T-1 line is first connected to a piece of equipment called the channel service unit (CSU). The CSU used to be considered part of the network, but is now almost always customerprovided and may also be merely included as an adjunct section in a complete T-1 interface solution. The CSU contains the last signal regenerator as well as a number of testing and maintenance features such as provision for loopback testing by the central office. It may also include a system to collect and report error statistics. The data service unit (DSU) handles the remaining digital housekeeping functions and data conversion from the bipolar T-1 format to standard serial data

  • The Multiplexer and Channel Cards. The multiplexer, sometimes called a channel bank, is where the multiple voice (or data) channels are combined into the single bit stream required for T-1 transmission. Each voice channel is converted to and from digital using codecs. In order to simulate typical telco lines, talk battery is added, ringing voltage is generated and loop current is detected. Generally, multiplexers are constructed using a modular circuit card approach so that the available digital bandwidth may be configured as desired.

Many modern PBX systems and at least one broadcast on-air system are able to accept T-1 lines directly. This is a near ideal approach, since you get a low cost direct digital connection into the telco network. In this case, no multiplexer and channel cards are necessary, because the connection is made directly to the CSU/ DSU. Some PBX equipment even incorporates the DSU.

T-1 and the Broadcast Interface

Generally, T-1 service appears to be a good idea for broadcasters, and many stations are using it successfully. However, be aware that some T-1 terminal equipment has problems in its analog conversion section, which cause the on-air hybrid interface to work very poorly with bad cancellation the result. Also keep in mind that, since all of your service will depend upon a single set of circuits, reliability could be reduced compared to individual analog lines. Consider having back-up circuits in place.

Primary Rate ISDN (PRI)

Primary rate ISDN has a data rate equivalent to T-1 circuits, providing 23B`D, or 23 64 kbps bearer channels and a 64 kbps D channel for control. (In Europe, PRIs have 31 bearer channels.) It is expected to replace T-1 eventually, since it speeds dialing and offers superior monitoring capabilities.

ADSL

Asymmetric digital subscriber lines (ADSL) promise connections at speeds of up to 3 mbps in the direction from the CO to the user. The upstream speed is limited to some much smaller value which is where the asymmetric part of the name comes from. An important advantage is the cost; it appears that this service may be priced at around the same level as ISDN BRI.

Initially, this technology was viewed by the telco industry as a way to compete with cable TV for the delivery of video services. Combined with an MPEG video/audio encoder, the bit rate offered by ADSL would permit full-quality National Television System Committee (NTSC) television. These projects now appear to be stalled and current efforts are being focused on high speed Internet connectivity. Since the Internet is a packet-based system with no bandwidth guarantee, the utility of this service for broadcast audio transmission is unclear.

t1-connector

Centrex

This service goes by various names, but the consistent principle is that the telco’s CO equipment replaces customer-owned PBXs. Each phone set has a direct connection to the CO. The idea is to eliminate customer up-front costs and transfer maintenance responsibility to the telco. Varying requirements for numbers of lines or phones can be accommodated without customer equipment upgrades. Centrex is declining in popularity but seems to remain popular with universities.

Features in Centrex rely upon flashing the switchhook and the use of the normal dialpad keys, generally an awkward and confusing situation for users. This problem may be solved with ISDN Centrex, as this permits very sophisticated phones to be used with all of the usual PBX features.

Cellular Telephone

Cellular extends the dial-up network to many places where a wire connection would not be considered practical. Cellular transceivers operate in the 800 MHz range and automatically select the appropriate frequency from among the 666 FM channels assigned for this service. Low power is used so that the frequencies can be re-used in adjacent areas. The mobile phone varies its power according to the level of signal received at the base location. A useful feature for onair use of a cellular phone is the signal strength meter provided on some units. Some phones also allow you to see the send power value. Often, the antenna’s pattern is quite directional due to its position on the vehicle, so moving around while observing the level indication can help make remotes sound better. For fixed remotes, a Yagi antenna can be used with its benefits of higher gain and directionality. At 800 MHz, Yagis are very compact.

Most equipment designed for use with wired phone lines can be connected to cellular phones using an adapter provided by the phone manufacturer. Intended for laptop computer modems and portable fax machines, these adapters provide an interface to any broadcast equipment that can connect to a phone line. Units especially designed for broadcast use have provisions for audio input and output for direct connection to microphone mixers and the like.

Some new digital cellular systems have the capability to transfer data via a special interface. Unfortunately, the bit rate is limited to only 14.4 kbps—not sufficiently fast for digitized audio. The impetus from the Internet may cause cellular vendors to offer higher bit rate phones in the future, permitting broadcasters to use them for high-fidelity remotes.

A downside of the new digital phones is that speech quality may be poor. This results from the very low bit rate used by these systems and the extreme compression methods that are required to shoehorn audio into the channel.

FCC Regulations

FCC requirements for connecting equipment to phone lines are outlined in Title 47 of the Code of Federal Regulations (CFR), Part 68: Connection of Terminal Equipment to the Telephone Network. The CFR can be ordered from the Government Printing Office.

 

PBX AND KEY SYSTEMS

Now that we know a bit about the nature of the phone network, we can explore what happens after the lines become ours. We will want to use some of what the phone people refer to as CPE. That is all of the equipment connected to the phone line after the official demarcation point. We will survey the various styles of PBX systems available both for general office and on-air use, followed by a look at systems designed specifically for studio application.

Private branch exchanges (PBXs) are found where there is a need for a large number of extension phones. PBXs are miniature central office exchanges, allowing local phones to call each other as well as access trunk lines for incoming and outgoing calls. PBX systems often have a number of specialized features for call routing and control. Traditionally, PBX systems have used only single-line phone sets as terminals, with special functions like transferring and conferencing accessible by flashing the switch hook or by using the tone pad in a special way. Most PBXs now have available feature phones, which can button-access individual lines as well as provide numerous other advanced functions. Sometimes these systems are called key systems after the old multi-key 1A2 phones. (Why phone engineers called buttons keys remains a mystery.)

Modern Telephone Systems

While the systems are tremendously varied, most have in common that the cable from each phone set to the common equipment conveys:

  • Power to operate the phone

  • A two-way data path to signal user actions from the set to the switch and operational and display status from the switch to the set x The speech audio.

Here are the usual approaches phone manufacturers employ for wiring and communication:

  • All Digital. The most advanced systems use a pure digital bit stream for both voice and data. The phone set contains the codec for conversion to-and-from the analog and digital domains. The pure digital approach is used in the AT&T System 85, in the Northern Telecom Meridian family, in the newer Mitel systems with the Superset DN phones and in the digital version of the NEC NEAX, among many others. The Siemens Office Point system claims to use standard ISDN protocol between the sets and the common equipment

  • Separate Pair per Function. The early electronic phones used a separate pair for each of the three functions, and thus required three (or more) pairs. The AT&T Merlin system used this design. The center pair is the audio; another pair is for the serially transmitted control and display data and another handles the phone’s power requirements

  • Two-Pair, Phantom Power. This used to be the most common approach, but is now fading, as pure digital designs have become cost-effective. The AT&T Spirit system the popular NEC and TIE systems and many others use this approach. Talk and data each use one of the two pairs. The power is applied between the two pairs similar to the method used for phantom powering condenser microphones in recording studios. A transformer at each end of the audio pair permits the phantom power to be added. The data pair will probably use resistors to obtain a center tap, rather than transformers since the data signal has a dc component which could not pass through a transformer.

  • Two-Pair, Power not Phantom. Some two-pair systems put the data on one pair and the audio on the other. Power may be on the data pair or on the audio pair. In the latter case, the audio pair resembles a central office line so that the phone ports may be universal: either single-line sets or feature phones can be plugged-in without hardware changes in the PBX. At least one of the Panasonic systems uses this technique. The center pair, again, is generally the audio

  • Data Over Voice. The analog Mitel Superset phones use a unique scheme that requires only one pair for all three functions. The data is amplitude shift modulated onto a 32 kHz carrier over voice and then the combined voice and data are ac coupled across the dc power voltage.

Interfacing to PBX Phones

It is usually possible to interface to PBX phones for on-air use. However, this is best reserved for casual phone use such as for the occasional request or contest winner call. For applications where phone calls are a significant programming element, it is usually better to consider the specialized on-air systems from the broadcast-oriented manufacturers.

One reason is that the hybrid interface cannot determine when a new call is selected, so it can not adjust its null to the new line before the conversation starts. (However, since the hybrid can null on voice during conversation, null will be achieved in perhaps four seconds. This is acceptable if only a portion of the call is to be aired, as is common with on-air requests, contest winner calls and the like.) Another shortcoming of the direct-to-electronic phone approach is that the line switching clunk is not muted, although this is not a problem when calls are not aired directly and sequentially.

Another potential problem is audio quality. The primary impediment is usually noise, most often the result of the data signals cross-talking into the audio. Buzz from the power supply sometimes finds its way into the audio. Often, frequency response is limited by too small line coupling transformers or from other causes. Poorly designed digital systems may suffer from quantization and aliasing noise and distortion. Few PBX manufacturers publish specs on audio performance. Since, clearly, this is of importance to those of us who need to get decent quality from phones for on-air use, we’ll want to make sure that the audio is at least reasonable. When choosing a new PBX, ask the phone system dealer for audio performance data or arrange to conduct at least a few simple tests yourself.

Direct Connection to the Skinny Wire

When the phone system uses the separate-pair approach previously described, the center two wires on the modular plug are usually the audio path. Since the phone’s control functions stay active even when these connections are broken, it is possible to intercept the audio signal here for feed to the interface. Most broadcast interfaces provide a loop-through connection, which feeds the phone line back out when it’s not active. Thus, the unit may be series connected with the audio pair. That way, you have normal telephone function preserved when the interface is not in use. When the interface is active, the phone serves merely as a controller, with no audio reaching the phone’s network or handset. Wiring the hybrid’s on/off functions to the console’s switching logic accomplishes automatic operation.

When the phone uses the two-pair phantom approach previously described, the audio is again likely to be present on the inner pair and may be intercepted for interfacing use if the dc connection is maintained. One way to do this is to provide a bypass for dc with inductors. Two mH has proven acceptable in experiments performed on some phone systems As shown in Figure 3.10-12.

fig-3.10

Special System Ports: Faux CO Lines

Since fax machines and modems need connections that look like central office lines, many systems provide ports for this use. They may be connected to broadcast interfacing gear as if they were CO lines. Sophisticated PBXs have programming features that allow these ports to be configured in various and potentially useful ways. For example, they may be set up for private line ringing (when a given incoming CO line rings, the call may be directly sent to the selected port). Unfortunately, with most PBX systems, awkward operation may result, since the only way to move a call from a phone set to the port may be to transfer it using multiple button punches, rather than the usual simple place-on-hold-and-pick-up-elsewhere operation. Taking calls in sequence on-air may be extremely difficult or impossible. Figure 3.1013 illustrates one possible solution.

Speakerphone Tap-Off

One way to get low cost interfacing is to take advantage of the switching-type interface that many phone set internal speakerphones provide. The procedure is to tap off the speaker with a transformer and pad to the console’s required input level. You may continue to use the phone’s internal microphone or you can provide an external send audio source to substitute for the phone’s internal microphone. Again, you will certainly need a pad and probably a transformer. The input feed must be set so that appropriate switching action and proper send levels are obtained.

Handset Adapters

Adapters are available that plug into the phone set’s handset modular jack and convert the microphone and earpiece signals into a signal that emulates a standard CO line. While useful in some applications, this approach is likely to offer a lower quality feed because the phone set’s network remains in the signal path causing impedance bumps and other problems.

Intercepting the Serial Data Stream

Why can’t we just emulate an electronic phone set by generating and decoding the phone system’s serial data? It does seem that this would be a good solution. However, phone system manufacturers insist on keeping their data protocols a deep secret. That means that broadcast manufacturers are unable to design direct emulation equipment. Of course, even if we had the protocols, there is the problem of accommodating the dozens of communication methods employed by PBX designers.

1A2 Key Systems

While nearly all stations have gone to high-tech PBXs for the business office, many on-air installations continue to rely upon 1A2 key systems. Key systems offer the advantage of providing a direct metallic connection to the CO line. That means that no frequency response error, noise, distortion or time delay is introduced. Often, these issues are not fully considered in the design of the more complex business phone systems. In addition, costs are favorable, and full schematics and other documentation are readily available.

Leading from the key service unit (KSU) to each phone is a thick cable with 50 conductors (25 pairs). The tip/ring pair carries the telephone audio. As mentioned, these are direct connections to the telco CO lines. The A leads tell the key system which lines are in use and also signal a hold condition. Selecting a line causes a connection to be made in the phone set from the A lead to another wire, the A-common. The A lead is normally at 124 Vdc and A-common is at ground potential, so when a line is selected, the A lead goes from124 Vdc to ground. If the A lead is broken before the tip/ring is disconnected, the system puts the line on hold. The lamp-leads light the phone’s line buttons with 10 Vac from the KSU’s power supply and are returned via the lamp grounds. The standard color codes and pinout are given in Table 3.10-2.

on-air-system

tele-color-code

The Evolving Phone

As time goes on, probably all but the most inexpensive systems will use the purely digital approach. As we’ve seen, these systems are difficult to interface to, but perhaps over time protocols will become standardized and maybe even based on ISDN. If this happens, broadcast interface manufacturers may be able to provide equipment that could directly connect to the PBX in place of, or in series with, the studio phone set.

Computer Telephony Integration (CTI)

With such a system, the PBX manufacturer provides complete documentation on an interface that can provide control of all of the important aspects of phone switching, including call set-up and routing functions. A standard data port is provided so that outside vendors may supply systems to work in concert with the phone equipment. These open PBXs may eventually offer a universal method for broadcast equipment to coordinate with the station’s office phone system.

Another approach is to build a PBX using special cards and software installed in a standard PC. Systems of this type would use the Windows NT operating system along with other standard PC software components such as a database server to provide a very sophisticated package of features. It is possible, for instance, to dial using database name look-up on a networked PC. Ironically, most CTI systems use analog phone sets.

 

BROADCAST INTERFACING

This section describes the techniques necessary to achieve the best possible result from the phone-tobroadcast shotgun marriage.

One-way Interfacing

There is often a need to take audio from a phone or broadcast in only one direction at a time (newsroom phoners are a common application). If there is no requirement for a two-way conversation, a simple interface using a QKT will do. Formerly available from the phone company, this small box was permanently wired into a phone instrument or line and provided a quarter-inch (12.7 mm) phone jack output for feeding a line-level signal to a console or recorder input.

Since the QKT is nothing more than a transformer, a capacitor and a zener diode limiter, you can make your own (see Figure 3.10-14). The capacitor provides dc blocking so that the transformer does not become saturated with the phone line’s dc potential. In order for the coupler to hold the line by drawing loop current, eliminate the capacitor and use a transformer that can withstand the loop current without producing distortion. (One such a transformer is the SPT117 from Prem Magnetics.) When sending audio into the phone line, remember audio level should be limited to 19 dBm. The QKT had back-to-back zeners for this purpose; you may want to add them to your homemade interface if you expect audio levels to get out of hand. Of course, commercial units are available that are a little fancier than the simple device described here. Some offer auto-answer and disconnect capability.

When using a coupler, it is most convenient to have the telephone instrument on-line and equipped with a push-to-talk switch on its receiver. This is because the phone’s receiver has to be off-hook while a feed is coming in; the switch turns off the receiver’s mouthpiece microphone when it is not depressed, thus insuring that noise from the studio side will not be included in the recording. Since this coupler works in both directions, it can be used to send audio down the phone as well—useful in the production studio for letting clients hear their commercial masterpieces before they go into the control room.

1way-interface

switching-interface

When hooking up to a multi-line phone, connect to a point where the tip/ring is present after line selection. The most convenient place is usually right at the phone network. Use headphones to find the spot.

Two-way Interfacing

The simple coupler’s limitations become apparent when it is necessary for the caller to hear the announcer and the audience to hear the caller simultaneously. A more sophisticated method is needed because of the requirement to have isolated send and receive audio signals.

Switching

This is what you get when you connect a speakerphone to your console input. No commercial broadcast interface uses this technique, which uses gain switching to keep the send audio from appearing at the receive output. Two electronic switches or voltage controlled amplifiers are used in such a way as to ensure that either the send or the receive path is closed at any given time, but never both simultaneously (see Figure 3.10-15). A decision circuit compares the send and receive levels, with the direction of transmission being determined by the relative signal strengths.

The disadvantage of the switching technique is its uni-directional nature. The caller cannot be heard while the announcer is speaking, and noises in the studio can sometimes cause a caller to disappear momentarily, especially on weak calls.

The Hybrid

Hybrids were invented long ago to separate the send and receive signals from the common two-way phone pair. Early hybrids were made from transformers with multiple windings. Nowadays, most hybrids are made with active components and are known as active hybrids. Both circuit types use the same principle and achieve the same effect.

In Figure 3.10-16, the first op-amp is simply a buffer. The second is used as a differential amplifier; the two inputs are added out-of-phase (subtracted). If the phone lines and the balancing network have identical characteristics, then the send signals at the second differential amp will be identical, and no send audio will appear at the output.

op-amp-hybrid

The balancing network is a circuit consisting of capacitance, resistance and sometimes inductance, forming an impedance network. Depending on the hybrid’s application, this circuit can be very simple or it can be comprised of a large number of components and have a very complex impedance characteristic.

R1 and the phone line form a voltage divider, as does R2 and the balancing network. If the phone line and balancing network are pure resistances, then, clearly, the phone line and the balancing network must have the same value in order for the signals at the differential amplifier to have the same amplitude and for complete cancellation to occur.

The phone line, however, is not purely resistive, but rather is complex impedance, causing both the amplitude and phase to vary as the send signal frequency varies. Two-to-four wire converters, transformers, repeaters, T-carrier systems and other telco systems are responsible for significant impedance bumps. Loading coils also usually have a deleterious effect on the performance of hybrid interfaces since the coils can create resonant peaks and phase anomalies in the phone line’s impedance curve which are difficult to null out.

Only when the impedance of the balancing network is the same as the phone line, and the signals at the differential amplifier are matched in both amplitude and phase, will full cancellation of the send signal be achieved. Otherwise, leakage results—the scourge of hybrids.

Because the phone company’s requirements are not generally too stringent, they usually use a simp



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