Digital Multiplexing - Statistical Time Division Multiplexing

In my last legacy Public Switched Telephone Network (PSTN) post I covered Time Division Multiplexing (TDM). I described how TDM works and why it does not efficiently use bandwidth. In this post let's take a look at Statistical Time Division Multiplexing (STDM or STATDM or STAT MUX), a much more efficient way to multiplex.

A Statistical Time Division Multiplexer (STDM or STATDM or STAT MUX) does not assign specific time slots for each device. An STDM adds an address field to each time slot in the frame and does not transmit empty frames. Only devices that require time slots get them. 

STDM uses dynamic time slot lengths that are variable. Communicating devices that are very active will be assigned greater time slot lengths than devices that are less active. If a device is idle, it will not receive any time slots. For periods where there is much activity STDMs have buffer memory for temporary data storage. 

STDM Multiplexing

Each STDM transmission carries channel identifier information. Channel identifier information includes source device address and a count of the number of data characters that belong to the listed source address. Channel identifiers are extra and considered overhead and are not data.  To reduce the cost of channel identifier overhead it makes sense to group large numbers of characters for each channel together.

In my next legacy PSTN post I'll cover Wavelength Division Multiplexing (WDM).

Digital Multiplexing - Time Division Multiplexing

In my last legacy Public Switched Telephone Network (PSTN) post I covered analog or frequency multiplexing. Frequency division multiplexing is now considered obsolete technology on the telecommunications network. Analog signals are more sensitive to noise and other signals which can cause problems along the transmission path. They have been replaced with digital multiplexers. 

Digital signals are combined or multiplexed typically using one of two techniques; Time Division Multiplexing (TDM) and Statistical Time Division Multiplexing (STDM). Let's cover TDM in this post.

Time Division Multiplexing allows multiple devices to communicate over the same circuit by assigning time slots for each device on the line. Devices communicating using TDM are typically placed in groups that are multiples of 4.

Each device is assigned a time slot where the TDM will accept an 8 bit character from the device. A TDM frame is then built and transmitted over the circuit. Another TDM on the other end of the circuit de-multiplexes the frame.

TDM Framing

TDM’s tend to waste time slots because a time slot is allocated for each device regardless of whether that device has anything to send. For example, in a TDM system if only two of four devices want to send and use frame space, the other two devices will not have anything to send.

TDM Framing Showing Wasted Slots

They do not require frame space but their time slot is still allocated and will be transmitted as empty frames. This is not an efficient use of bandwidth.

In my next legacy PSTN post, I'll cover statistical time division multiplexing (STDM), a much more efficient way to use bandwidth.

Analog or Frequency Multiplexing

In this post I continue discussing some of the different legacy technologies used by the Public Switched Telephone Network (PSTN). Today let's take a dive into analog or frequency multiplexing.

Analog or frequency multiplexing is now an obsolete technology in the U.S. telecommunications industry. It was used up until the early 1990’s by long-distance carriers like AT&T and MCI and is still used today in other countries. The concept of channel banks was developed for analog multiplexing and this concept is still used today for other types of multiplexing. To multiplex calls each call was given a narrow range of frequency in the available bandwidth. We know all voice call channels occupy the same frequency range – approximately 4000 Hz if we include individual call guardbands. If we want to combine a group of voice calls and separate them by frequency we must translate the frequency of these individual call channels using a process called Single Sideband, Suppressed Carrier Modulation. This technique allowed 10,800 individual voice call channels to be combined and transmitted over one coaxial copper pair. Let’s look at how it was done.

Groups

Individual voice call channels are placed into groups of 12. If we have 12 channels per group and each channel is 4000 Hz we can calculate:

This 48KHz is placed in the frequency range of 60 – 108 KHz. 

Single Group Formation

Supergroups

Individual groups are placed into supergroups of 5 and each supergroup contains 60 individual voice channels. If we have 5 groups and each group is 48 KHz we can calculate:

This 240KHz is placed in the frequency range of 312 – 552 KHz.

Supergroup Formation

Mastergroups

Individual supergroups are placed into mastergroups of 10 and each mastergroup contains 600 individual voice channels. If we have 10 supergroups and each supergroup is 240 KHz we can calculate:

This 2.40MHz is placed in the frequency range of 564 – 2.964 MHz.

Mastergroup Formation

Jumbogroups

Individual mastergroups are placed into jumbogroups of 6 and each jumbogroup contains 3600 individual voice channels. If we have 6 mastergroups and each mastergroup is 2.4 MHz we can calculate:

This 14.4 MHz is placed in the frequency range of 3.084 – 17.484 MHz.

Jumbogroup Formation

Jumbogroup Multiplex

The final multiplexing step involves combining individual jumbogroups which are placed into jumbogroup multiplexes of 3. Each jumbogroup multiplex contains 10,800 individual voice channels. I'm still amazed - 10,800 calls on one piece of coaxial cable!

Frequency multiplexing is now considered obsolete technology on the telecommunications network. Analog signals are more sensitive to noise and other signals which can cause problems along the transmission path. Those long coaxial cables make pretty good antennas. They have been replaced with digital multiplexers. In my next legacy PSTN post I'll cover how digital multiplexing works.

reference: Introduction to Telecommunications Networks by Gordon F Snyder Jr, 2002

Multiplexing - A Brief Introduction

In this post I continue discussing some of the different legacy technologies used by the Public Switched Telephone Network (PSTN). Today let's take a quick look at what multiplexing is.

Before the invention of the telephone both Alexander Graham Bell and Thomas Edison were experimenting with ways to transmit more than one telegraph signal at a time over a single wire. They both realized this was a critical piece if any communications network was to grow in the number of users.

Multiplexing

There are three ways to multiplex or combine multiple signals on the telephone network. They are analog or frequency multiplexing, digital multiplexing and wavelength division multiplexing. I'll dig pretty deep into each in upcoming legacy posts.

The SLC-96

In this post I continue discussing some of the different legacy technologies used by the Public Switched Telephone Network (PSTN). Today let's take a look at how the PSTN designed and tuned for voice communications started to change in the late 1970's with something called an SLC-96 (pronounced "Slick 96").

It's still not economical even today to run fiber into every home but Local Exchange Carriers like Verizon and AT&T have been working to replace portions of the local loop with fiber by running fiber out from the CO into a Remote Terminal (RT) pedestal box in the field called a Multiple Subscriber Line (or Loop) Carrier System or SLC-96. Each SLC-96 takes 96 64 Kbps analog voice or modem signals, converts them to digital and then multiplexes them at the Remote Terminal. The Remote Terminal is connected to a Central Office Terminal (COT) using 5 T1 (DS-1) lines. 

SLC-96 Field Pedestal Configuration

Four of these T1 lines are used to carry the 96 digitized voice channels (1 T1 line = 24 digitized voice channels so 4 T1’s are required to transmit 96 voice channels). The fifth T1 line is used for protective switching and is a backup if one of the four fails.

In my next legacy PSTN post I'll start covering multiplexing.

A Few Additional Telephone System Features

In this post I continue to describe the legacy Public Switched Telephone Network (PSTN), looking at a few other common telephone system features we are all used to having and relying on. These are additional handset signals and PIC. I would also want to include Caller ID here but I've already covered how it works in a previous post.

Some Common Handset Signals

We are all used to hearing these additional common signals coming from our telephone. 

Line Busy Signal - 480 Hz and 630 Hz tones on for .5 seconds and off for .5 seconds, then repeats. 

Block Signal - 480 Hz and 620 Hz tones on for .2 seconds and off for .3 seconds, then repeats. This signal is often referred to as fast busy.

Off-Hook - 1400 Hz, 2060 Hz, 2450 Hz and 2600 Hz tones on for .1 seconds and off for .1 seconds, then repeats with a duration of 40 seconds. This signal is designed to be heard from across a room and is very loud.

Preferred Interexchange Carrier (PIC)

Since the 1976 MCI ruling AT&T has been required to open the long distance market to other long distance providers. Prior to this, all long distance traffic in the United States was handled by AT&T and users would just dial a “1” to connect to an AT&T long distance trunk. As other long distance carriers entered the market, AT&T had a big advantage. Customers were already used to dialing a “1” for long distance and placing a long distance call to anywhere in the United States involved dialing a minimum number of numbers. – only 11. This included “1”, the area code, and the 7 digit number. Customers that wanted to used other long distance carriers had to dial 25 numbers to make a long distance call. These calls required an 800 number be called initially (11 numbers), a 4 number personal identification number (PIN), the area code, and the 7 digit number.

In 1987 a method called Feature Group D was implemented to automatically pass calls to the customers preferred long distance carrier using something called a Preferred Interexchange Carrier (PIC) number.  Customers are required to select a preferred carrier and the preferred carrier information is added to the local switch database the customer is connected to.

Feature Group D also allows a customer to bypass the preferred PIC by dialing a 101XXXX number and use another long distance carrier. These 101XXXX are commonly referred to as dial-around service numbers.

Telephone Set Function 4. To convert voice frequencies to electrical signals that can be transmitted

In my last few legacy Public Switched Telephone Network (PSTN) posts, I covered pulse or rotary dial service, dual tone multi frequency (DTMF) dialing service and what makes a telephone ring. In this post let's look at microphones and speakers.

A telephone converts voice frequencies to electrical signals and electrical signals back to voice frequencies using basic microphone transmitter and speaker theory and application. 

Transmitters

A telephone transmitter is built into the handset of the phone and is responsible for converting sound waves into electrical signals that can be transmitted.

Telephone Carbon Granule Transmitter

Carbon granule transmitters are still common in wired home phones. Sound travels in waves that are actually variations in air pressure. Some of these waves enter the mouthpiece and cause a diaphragm in the transmitter microphone to vibrate back and forth. These vibrations put either more or less pressure on carbon granules in the base of the microphone. If more pressure is applied, the granules pack more tightly and conduct electricity more efficiently. Inversely, in between the waves the granules unpack and do not conduct as well. Voltage is applied across the electrical contacts and the varying amounts of resistance caused by the carbon granules in the microphone cause varying amounts of current to flow. This current variation is an electrical representation of the sound waves (voice=analog signal) entering the microphone. 

In addition to carbon granule transmitters many modern telephones use dynamic transmitters that function by moving a coil of wire inside a magnetic field to produce an electrical current in response to soundwaves or electret transmitters, also known as condenser microphones, which use a capacitor for a transducer and generally contain an amplifier circuit. 

Receivers

The telephone handset receiver is just a simple speaker. It performs the opposite function of the transmitter in that it takes the incoming electrical signal and converts it to sound waves that can be heard by the listener. 

Simple Speaker Diagram

The incoming electrical signal flows through a magnetic coil in the speaker. The magnetic field surrounding the coil changes in conjunction with the changing current flowing through the coil. This changing magnetic field causes a cone in the speaker to vibrate. These vibrations create air pressure waves forming sound.

In my next legacy PSTN post I'll describe how some additional telephone features work.

Telephone Set Function 3 - To provide a way for the telephone company to indicate that a call is coming in or ringing

In my last two legacy Public Switched Telephone Network (PSTN) posts I covered pulse or rotary dial service along with dual tone multi frequency (DTMF) dialing service. In this post let's look at what makes a telephone ring.

When the user begins dialing the phone each sequenced number is stored in the central office computerized switch at the Local Exchange Carrier (LEC) Central Office (CO) and analyzed. The first three digits determine if the call is local or long distance. If the call is local, the switch determines if it can complete the call itself or the call needs to be forwarded to another local LEC CO that handles that telephone number. If the call is long distance, the call needs to be forwarded to the customer's long distance carrier.

Once the call destination is determined a switch on the receiving end sends a repeating 90 Vrms 20 Hz ringing signal (on for 2 seconds with a 4 second pause) called a ring or alerting signal to the receiving phone causing it to ring.

Ring or Alerting Signal

Notice the ringing signal has an inaudible frequency of 20 Hz - this is why different phones can have different ring styles.

At the same time, a ring back signal that is a mix of 440Hz and 480Hz is sent back to the caller. This signal is on for 2 seconds and off for 4 seconds and indicates that the phone being dialed is ringing. When the receiver picks up the handset the telephone goes off-hook.  The switch hook on the receiving phone closes, current flows and the CO switches turn off the ringing signals.

Telephone Set Function 2. To provide the telephone company with the number the caller wishes to call - Part 1

In this post I continue legacy Public Switched Telephone Network (PSTN) technology coverage.

There are two methods currently used to provide numbers to the telephone company, pulse or rotary dial service and dual tone multi frequency dialing. Let's look at pulse or rotary dial service in this post.

In the past, when a handset was lifted, the caller did not hear dialtone, the caller heard an operator asking for the number the caller wanted to dial. As the number of telephones grew, telephone companies projected that hundreds of thousands of new operators would be needed so rotary dials were added to telephones.

Rotary dials were invented to eliminate operators and use dial pulsing to automate the switching required to get from a caller to a receiver. The rotary dial generates pulses on the local loop by opening and closing an electrical switch when the dial is rotated and released. Each pulse opens the loop and interrupts the local loop current flow of 20 - 120mA resulting in a series of current pulses on the local loop. This process is referred to as out-pulsing and pulses are generated at a rate of ten pulses per second. Each pulse is actually an interruption in current flow on the loop and is .05s with a .05s pause between pulses. Each number on the dial corresponds to the number of pulses produced for that number. For example, dialing the number 4 produces four pulses as indicated in the figure below  and takes a total of .4 seconds (8 x ,05 seconds = ,4 seconds). As you can see, rotary dialers are slow when compared to modern telephones today.

Telephone Rotary Dial Timing Diagram of the Number “4”

Example

How long does it take to dial the single number "9" on a mechanical rotary phone?

Solution

Dialing the number "9" produces: 

.05s pulse, .05s pause, 05s pulse, .05s pause, 05s pulse, .05s pause, .05s pulse, .05s pause, 05s pulse, .05s pause, 05s pulse, .05s pause, .05s pulse, .05s pause, 05s pulse, .05s pause, 05s pulse, .05s pause

.05 seconds x 18 = .9 seconds

As telephone manufacturing technology developed the rotary dials were replaced on many phones with a push-button keypad. These keypads use an electronic circuit to generate the pulses, not a mechanical rotary dial. Since people can punch numbers very rapidly and pulse signals still must be .05s long and be separated by .05s pauses, this type of dial is equipped with a buffer that stores numbers as they are keyed. The buffer then out-pulses the numbers with the proper timing intervals. You may also have noticed a telephone "digital" keypad number sequence is opposite that of a calculator. This was done purposely to slow people down when dialing on pulse generators.

Pulse generation phones still work on the Public Switched Telephone Network (PSTN). It's amazing the telephone companies still support these now almost obsolete phones! In my next telephone technology post I'll cover dual tone multi frequency dialing.

Dialtone Generation

In my last telephone set post I described what happens when a telephone handset is picked up. Here's a diagram showing that process.

Dialtone Generation

Dialtone is a signal formed by the simultaneous transmission of a 330 Hz tone and a 440 Hz tone. After the first number is dialed by the caller the dial-tone generator is shut off.