Testimonios epigráficos

Figure 3.10-1.mLaw PCM coding within the telephone network causes the noise to be approximately a fixed percentage regard-less of level.

SECTION3: AUDIOPRODUCTIONFACILITIES

analog (D/A) integrated circuits called codecs (CODer/

DECoders). The method is specified by the Interna-tional Telecommunications Union (ITU) as standard G.711.

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. Non-copper 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 in-cluded 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 at148 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 1,300V depending on length. Because of this series resistance, when a line is off-hook, its voltage at the customer equipment drops to around112 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 sinus-oidal shape shifted 48 V to the negative.

Talk signals are ac coupled with nominal impedance of 600V. However, some CO equipment uses complex impedance coupling, and the nature of the telephone net-work usually results in the actual impedance as pre-sented to the user rarely being the specified simple 600V. This turns out to be an important issue for broad-cast 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

Table 3.10-1 Phone loop characteristics.

Parameter Typical U.S. Values Operating Limits Talk Battery Voltage 148 VDC 147 to1105 VDC

Loop Current 20 to 80 mA 20 to 120 mA

Loop Resistance O to 1300 ohms 0 to 3600 ohms

Loop Loss 8 dB 17 dB

Distortion 150 dB N.A.

Ringing Signal 20 Hz, 90 VRMS 16 to 60 Hz, 40 to 130 VRMS Noise (objective) 169 dBm0 to 180 mi,

150 dBm0 to 3000 mi (116 dBm0 talk level) (C msg weight)

434

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 434 Figure 3.10-2. Two-wire circuits have both directions on a single pair of wires, which are separated for switching and long-distance transmission into 4-wire signals with hybrids.

TELEPHONENETWORKINTERFACING

measuring up from there. The reference noise level is one picowatt, which corresponds to190 dBm. Thus, a noise level of160 dB relative to 0 dBm would be reported as 30 dBrn noise (dBrn4 dB 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 meas-uring phone line signal-to-noise ratio (S/N). (See Fig-ure 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 consider-able 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.

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 theoret-ical Nyquist frequency of 4 kHz, but a 600 Hz transition band is necessary for anti-aliasing and reconstruction filtering (see Figure 3.10-3).

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 perhaps140 to 14 dBm, as illustrated in Figure 3.10-4.

Send audio (that is, audio fed into the telephone line) must be limited to19 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

435

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 435 Figure 3.10-3. The low-pass filters required for digital transmission restrict frequency response. This response curve is for a codec that is widely used in the telephone network. (Note also the significant low frequency roll-off).

SECTION3: AUDIOPRODUCTIONFACILITIES

Figure 3.10-6. DTMF tone keypad frequency assignment. The four tone pairs in the last column are for special applications.

436

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 436 Figure 3.10-4. Signal and noise level references used in telephone engineering.

DTMF Tone Dialing

Dual Tone Multiple Frequency (DTMF) dialing uses two frequencies for each digit in order to avoid talk-off—that is, the tone detector accidentally sensing voice as a dial command. In addition, the frequencies were carefully chosen to avoid problems with har-monic distortion causing false detection. There are four

Figure 3.10-5. C-message weight frequency response curve.

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 and62% 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 between110

TELEPHONENETWORKINTERFACING

hangs up. Thus, we can use the presence of dial tone as a back up to cause a disconnect when the loop-current detection methods fail. An important consider-ation 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 num-ber 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 and 16 dBm; ideally, tones in the high group are

transmitted with 2 dB greater level in order to compen-sate for high-frequency roll-off in the phone line.

Loop Start and Ground Start

Central office lines come in two basic configura-tions: 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 termi-nal 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 con-nection 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 re-duced. 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 re-ferred 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 prolif-eration of answering machines that rely upon CPC, most central office equipment now has this capability designed in. In some cases, it is necessary to specifi-cally 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 eventu-ally returns dial tone to its lines when the calling party

437

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 437 Figure 3.10-7. Frequency response with and without loading coils.

SECTION3: AUDIOPRODUCTIONFACILITIES

separation in the two speech paths. They are not dial-up, 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 has-sles, so it will probably supplant these analog lines over time.

Foreign Exchange (FX) Loops

FX provides local telephone service from a central office that is outside (foreign to) the subscriber’s ex-change 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 interfac-ing. 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 pro-gram directors (PDs) desired the publicity that burning out a phone exchange would generate.

The choke network works by diverting calls begin-ning 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 serv-ing CO choke ports to the choke exchange. The usual switching and routing process is bypassed. Unfortu-nately, only a very limited number of paths are gener-ally 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 em-ployed 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 per-formance 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

438

National Assoc. of Broadcasters (NJ) (PS8295) PKF 01-06-99 09:34:51 CH3x10 Page 438 Figure 3.10-8. Typical choke network transmission path.

TELEPHONENETWORKINTERFACING

col 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 inte-grated 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 equip-ment to automatically adapt to various user require-ments by sensing different SPIDs from each type or configuration of user terminal. For instance, multi-button 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, connec-tion 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.

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.