Why the Public Switched Telephone Network Is Sunsetting

In my last post, titled Verizon No Longer Concerned With Telephones Connected With Wires, I described an interview Ivan Seidenberg, chief executive of Verizon Communications, did at a Goldman Sachs investor conference on Thursday. In the inteview Seidenberg described how, by using the decentralized structure of the Internet rather than the traditional design of phone systems, Verizon had a new opportunity to cut costs sharply.

This summer I spent some time reading Martin Sauter's excellent new book Beyond 3G, Bringing Networks, Terminals and the Web Together. In the book Martin describes the movement in the wireless/cellular world away from circuit-switched telephony technologies like 2G, 2.5G (EDGE) and even 3G to 4G based technologies like LTE and WiMAX.

What does wireless technology have to do with copper wires? Like these wireless technologies, the Public Switched Telephone Network (PSTN) uses circuit-switched telephony technology designed around voice. Even DSL (a technology basically designed to extend the life of the copper wire based network by a few years) is a circuit-switched service - Internet based traffic goes to the Internet and voice traffic goes - you guessed it - right to the PSTN.

Circuit-switch based networks have made a lot of sense for the past 100 years or so. They work well for voice calls because by nature they are deterministic. If a circuit is available a connection is made. If a circuit is not available the call attempt gets rejected and the customer gets some kind of message back from the busy switch. Once a connection is made (phone-to-phone) the connection is also deterministic - each call is independent and cannot influence any other calls. A great design for voice communications - whether it be with copper wires or over wireless frequencies.

The problem with these circuit-switch based networks though is they were designed for voice. Sauter argues correctly that when networks are designed for specific applications, there is no separation between the network and the applications which ultimately prevents evolution. In addition, tight integration of applications and networks also prevents the evolution of an application because changing the applications also requires changes to the network itself. The PSTN basically cannot evolve beyond where it is now - it's been tweaked-up to the point where it cannot be tweaked-up any more.

Internet (TCP/IP based) technologies work using exactly the opposite approach. A neutral transport layer carries packets and any kind of application (voice, video, data, etc) can efficiently send high and low volumes of data through the network. For applications the connection process is transparent - the device operating system establishes an Internet connection before the application is even launched. The network and any applications running that use the network are independent of each other.

Verizon Wireless, AT&T, Sprint, etc are all moving to non-circuit-switched IP based 4G technologies like WiMAX and LTE to handle voice, video and data traffic. It is inevitable that Verizon's landline division (along with other landline carriers) move in this same direction.

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.

Analog to Digital (and Digital to Analog) with CODECs

In this post I continue to discuss the (rapidly disappearing) Public Switched Telephone Network (PSTN).
CODECs are used to convert analog signals to digital signals on one end and, on the other end, convert a digital signal back to an analog signal.

 CODEC Conversions

CODECs use a method called Pulse Code Modulation (PCM) to convert the analog signals to digital bit streams. PCM uses a technique called sampling to obtain instantaneous voltage values at specific times in the analog signal cycle. This sample generates a Pulse Amplitude Modulated (PAM) signal.

PAM Signal Generation

The diagram above shows an analog signal multiplied with a digital pulse train instantaneous point by instantaneous point with the result being a PAM wave representation of the analog waveform. The digital pulse train determines the sampling rate and it is easy to see if the analog signal is not sampled enough, the analog signal will not be properly represented by the PAM signal. 

In 1924 while working for AT&T Henry Nyquist studied this sampling technique and developed the Nyquist Sampling Theorem. This theorem states that an analog signal can be uniquely reconstructed, without error, from samples taken at equal time intervals if the sampling rate is equal to, or greater than, twice the highest frequency component in the analog signal or:

Sampling Rate = 2(BW)

Example

The Public Switched Telephone Network (PSTN) has a bandwidth of approximately 3300 Hz. Using the Nyquist Sampling Theorem calculate the minimal sampling rate of the PSTN.

Solution: 

Sampling Rate = 2(BW) = 2(3300 Hz) = 6600 Samples per Second

This means, for minimal analog to digital conversion, an analog voice telephone line must be sampled minimally 6600 samples per second. Sampling at a rate of less than 6600 samples per second will not reproduce the signal properly.  Sampling rates of greater than 6600 samples per second will produce more detail. Designers of the voice network used Nyquist’s Sampling Theorem to determine the proper sampling rate. They knew they could not sample under the 6600 samples per second rate and also knew going over the 6600 samples per second rate would produce higher quality. A PCM sampling rate of 8000 samples per second was selected.

In my next post I'll discuss Quantization, which is used along with the sampling rate to generate a PCM wave.

Wavelength Division Multiplexing (WDM)

In my last legacy Public Switched Telephone Network (PSTN) post I covered Statistical Time Division Multiplexing (STDM).  In this post let's take a look at Wavelength Division Multiplexing (WDM and DWDM) methods.

As bandwidth requirements continue to grow for both the legacy Public Switched Telephone Network and the emerged Internet/IP network most of the high bandwidth backbone transmission is being done with fiber optics and a method called Wavelength Division Multiplexing or WDM. WDM functions very similarly to Frequency Division Multiplexing (FDM). With FDM different frequencies represent different communications channels with transmission done on copper or microwaves. WDM uses wavelength instead of frequency to differentiate the different communications channels.

Wavelength

Light is sinusoidal in nature and wavelength, represented by the Greek letter lambda (λ) is a distance measurement usually expressed in meters. Wavelength  is defined as the distance in meters of one sinusoidal cycle.

Wavelength Measurement

Wavelength indicates the color of light. For example, the human eye can see light ranging in frequency from approximately 380 nm (dark violet) to approximately 765 nm (red). WDM multiplexers use wavelength, or color, of light to combine signal channels onto a single piece of optical fiber. Each WDM signal is separated by wavelength “guardbands” to protect from signal crossover. One of WDM’s biggest advantages is that it allows incoming high bandwidth signal carriers that have already been multiplexed to be multiplexed together again and transmitted long distances over one piece of fiber.

Wavelength Division Multiplexing

In addition to WDM systems engineers have developed even higher capacity Dense Wavelength Division Multiplexing (DWDM) systems. Just this past week, Cisco and US Signal announced the successful completion of the first 100 Gigabit (100G) coherent DWDM trial. As backbone bandwidth requirements continue to grow these WDM and DWDM systems are significantly reducing long haul bandwidth bottlenecks.

More Telephone History (1878-1918)

A couple of weeks ago I pulled a piece out of a book I wrote about ten years ago titled Introduction to Telecommunications Networks. In that post I described the first year in the development of telephone technology. As a follow-up to that post, here's some of the major technical breakthroughs that happened between 1878 and 1918.

1878
Bell sets up the first operator switching exchange and at the same time, Western Union Telegraph Company (http://www.westernunion.com) decided to use its existing national telegraph wire network to set up its own telephone company. Bell quickly sued Western Union and Western Union settled out of court, selling its network to Bell.

Henry Hummings in England gets a British patent for a variable resistance telephone transmitter that used finely ground carbon. The carbon transmitter solved many of the early problems Bell had trying to use liquid and electromagnetic transmitters. The carbon transmitter also used a voice cone attached to a diaphragm.

The diaphragm, which was attached to a conductor, vibrated with sound waves and caused the closed container of ground carbon to compress and uncompress changing resistance in the same way the liquid transmitters did.

1885
American Telephone and Telegraph Company (http://www.att.com) was formed to provide long distance telephone service, connecting small Bell regional telephone franchises.

AT&T buys Henry Hummings’ ground carbon variable resistance telephone transmitter patent rights.

1886
Thomas Edison modified Henry Hummings’ finely ground carbon transmitter by using larger carbon granules. The larger granules created more current paths with sound wave compression and therefore allowed more current to flow in conjunction with the compression. The larger granules also did not pack as tightly over time like the finely ground carbon in Hummings’ transmitter. When they did pack, usually lightly hitting the transmitter on a hard surface would loosen them up.

1899
AT&T reorganizes, assuming the business and property of American Bell and becomes the parent company of the Bell System.

1908
Siemens (http://www.siemens.com) first tests dialtone on the public switched telephone network in a German city.

1918
AT&T patents an anti-sidetone solution for telephone receiver and transmitters. This technology allowed talkers to more easily adjust their voice volume when speaking into the telephone transmitter.

I'll continue with more history in a later post.

The Basic Telephone Set Fundamental Functions

With my recent posts on the Public Switched Telephone Network (PSTN) I've been getting some email questions and suggested posts. I've received a few questions on telephones (what I would call  end user devices) so I thought I'd take a few posts to describe how a basic telephone works.

The basic telephone set connected to the telephone network we are all very comfortable with using, has 4 basic functions:

1. To provide a signal to the telephone company that a call is to be made (off-hook) or a call is complete (on-hook). 
2. To provide the telephone company with the number the caller wishes to call.
3. To provide a way for the telephone company to indicate that a call is coming in or ringing.
4. To convert voice frequencies to electrical signals that can be transmitted at the transmitter and convert those electrical signals back to voice frequencies at the receiver.

The Federal Communications Commission (FCC) has set standards for the above features and all manufacturers selling telephones in this country must match these standards or the phone will not work properly. In addition many modern telephones also come with features like speed dial, redial, memory, caller ID, voice mail, etc. These are all additional features that are not necessary to make or receive calls.

Let's look at Telephone Set Function 1: To provide a signal to the telephone company that a call is to be made (off-hook) or a call is complete (on-hook).

The switchhook gets its name from the old telephones that had a hook on the side. On modern phones the switchhook is a button that is depressed when the handset is put on the cradle of the telephone. 

According to telephone company specifications individual telephone set DC resistance should be 200 Ω but in reality most telephones range between 150 and 1000 Ω of DC resistance. When a user picks up a connected telephone handset to make a call the switchhooks in the figure below (S1 and S2) close (off-hook condition) and the local loop circuit is complete. 

When a handset is picked up, a DC current ranging between 20 and 120 mA flows on the pair of wires connecting the telephone to the CO. This current flow causes a relay coil to magnetize and it's contacts close.

In the CO current flows through a relay coil attached to the local loop wire pair. The coil energizes, it’s contacts close and the CO switch knows a phone is off hook somewhere. A line feeder in the CO switch looks for the off-hook signal, finds it and sets up a connection. In the CO switch a dial-tone generator is connected to the line so the caller knows they can dial a number. 

I'll cover dial-tone generation (and why cell phones don't use dial-tone) in my next post.

Wireless and VoIP Services as Carrier of Last Resort?

The shift continues for the traditional telecommunications companies away from copper based voice and DSL data services to wireless and fiber. One of the road blocks that appears to be loosening are the  Carrier of Last Resort (COLR) rules for carriers.

COLR rules are currently set at the state level (not the Federal Communications Commission) and regulate that every American has access to telephones service along with other utilities like electricity and water. A number of states have either passed legislation or are considering legislation that would end traditional landline rules and allow these services to be replaced by wireless (cell) or Voice over Internet Protocol (VoIP) services. Bills have emerged in Mississippi, Kentucky, New Jersey and California. Ohio's Senate Bill 271 is a good example of legislation currently being reviewed by lawmakers to cut traditional landline services. 

Opponents to these changes argue landline elimination could increase phone bills, reduce quality of service and impact 911 service. AARP Ohio State Director Bill Sundenmeyer is quoted in a recent post at Community Broadband Networks saying:

... besides preserving social contact, land-line phones are needed to protect seniors' health and safety. For instance, some seniors use the phone line to transmit routine health information from equipment in their home to their doctor's office.They can make an evaluation of a person's heart and how's it working, of their lungs, etc. That information would be very difficult to transmit over a cell phone.

There's more. Even though the FCC has stayed out of COLR regulations, leaving them to individual states, AT&T submitted a letter to the FCC back in August asking the FCC to effectively reclassify the public switched telephone network as an "information service", effectively removing all PSTN regulations and obligations. What does this mean? I think Bruce Kushnick describes it pretty well over at the Huff Post Tech Blog:

This means that almost all of the remaining wires, networks or even the obligation to offer services over those wires and networks are all removed -- as much of this infrastructure is classified as "telecommunications". The Public Switched Telephone Networks, the utility, would suddenly be reclassified as an information service. Sayonara any telco rules, regulations and oh yes, your rights. Your service breaks... tough. Prices go up and there's no direct competition -- too bad. Networks weren't upgraded -- so what. Net Neutrality? Neutered.

I'm not sure where you live but I'm in a relatively rural area of a fairly populated state. I've only got one wireless provider option at my home unless I climb up to the very peak of my roof where I can usually catch one bar of another provider. After the 2011 Halloween snowstorm cell service was out for almost a week at my home while landline service did not go down. 

Wireless service is great when it works. Wireless as carrier of last resort - someday yes but not just yet. AT&T has opened a window and the FCC now has an opportunity to step up and put a logical transitional process in place. 

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.

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.

Those Copper Wires Coming Into Your House - The Local Loop

It’s all going to be going away soon but, for most of us, our landline phones are still connected the way they were 80 years ago......

The analog Public Switched Telephone Network (PSTN) or Plain Old Telephone Service (POTS) local loop is defined as the twisted pair of copper wires many of us have coming into our home or business. This local loop is sometimes referred to as the “final three miles” or simply the “final mile”. The local loop has been “tuned” to our voice frequencies over the last 100 years and has a bandwidth of approximately 4000 Hz. This bandwidth includes two guardbands to prevent adjacent frequency interference. As can be seen in the figure below, bandwidth available to the local loop circuit for actual voice analog transmission is about 3000 Hz.

PSTN Bandwidth

The local loop wire pair consists of two wires and runs from a home or business to a Local Exchange Carrier (LEC) Central Office (CO) which is also referred to as the Central Exchange (CE).  The CO provides voltage (– 48V DC) for the telephone in our homes and businesses. The wires that make up a wire pair are identified as follows: The “tip” (red wire) is attached to the negative side of the CO 48 V battery and the “ring” (green wire) is attached to the positive side of the CO 48 V battery.

Local Loop Telephone Circuit

This diagram shows a basic local loop telephone circuit. Notice the CO provides the voltage for the telephone. This voltage is provided by batteries in the CO – we’ve all experienced power failures at one time or another and most realize telephones still work even when the power is out. Also notice the battery polarity is inverted and a –48 V DC is being provided to the phone. This is done for electrolytic corrosion reasons. In my next post we’ll look at the local loop in the form of a transmission line.

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

In my last legacy Public Switched Telephone Network (PSTN) post I covered pulse or rotary dial service.  Let's look at dual tone multi frequency (DTMF) dialing service in this post.

The most commonly used method for inputting a number in the US and Europe is now the dual-tone-multifrequency (DTMF) signaling method. DTMF telephones are also commonly known as Touchtone telephones. These phones also use numerical keypads but offer an even faster way to signal the number to call by sending tones on the telephone line. The DTMF phone uses a 12-button keypad. When a button is pressed on the keypad an electric contact is closed and two oscillators generate two tones at specific frequencies. 

Telephone DTMF Keypad

These tones combine to form one sound to the listener, just like when two different musical notes on an instrument are played at exactly the same time. The combined tones are a signal for the button that was pressed on the keypad. The frequencies used are illustrated in the keypad diagram. For example, notice when the number 8 is pressed the frequencies 852 Hz and 1336 Hz are combined to form the number 8 tone. 

For the central office to accept tones from a caller, the tones must be at least 50 milliseconds long and also be separated by  a 50 millisecond pause. DTMF phones offer much more rapid dialing of numbers than rotary pulse methods with the average phone number taking 10 to 15 times less time to dial using a Touchtone phone. Not only are Touchtone phones faster, they are also more reliable because they do not depend on as many moving parts as a rotary phone.

In my next legacy PSTN post, I'll describe how a telephone is made to ring.

The First Year Of The Telephone

About ten years ago I wrote a book titled Introduction to Telecommunications Networks. About half the book described how the now rapidly disappearing public switched telephone network (PSTN) worked. I haven't picked up the book in a while but a recent flip through has certainly brought back some memories. I thought it would be interesting to take a look at some of the history. Here's how it all started.

1876
Alexander Graham Bell and Elisha Gray, another inventor competing with Bell, are both scrambling to get their voice transmission inventions patented.
 
February 14, 1876
On this day Alexander Graham Bell’s father in law, attorney Gardiner Hubbard, delivered a patent application from Bell to the U.S. Patent for a device that transmits voice frequencies across wires.

Approximately three hours later on the same day Elisha Gray filed a caveat (a formal notice of an invention Gray hoped to patent) with the U.S. Patent Office describing a device that also transmitted voice frequencies across wires.

March 10, 1876
Alexander Graham Bell and Thomas A. Watson demonstrate a working telephone system but not without controversy. When Bell’s original patent and Gray’s caveat, both filed on February 14, were reviewed it was determined the device Bell described would not have worked while Gray’s would have. It was speculated that Bell had copied parts of Gray’s design. In Gray’s caveat he had detailed the use of a variable resistance transmitter which was used to produce a transmitter signal robust enough for the receiver to hear. Bell had been struggling to solve this same problem. In Bell’s patent application he made what appeared to be a last minute handwritten notation about the use of a variable resistance transmitter. People speculated that Bell had found out about Gray’s caveat and learned of Gray’s use of a variable resistance transmitter and, at the last minute before filing, Bell made a note on the patent application about using the new transmitter.

The variable resistance transmitter demonstrated by Bell on March 10, 1876 used a voice cone attached to a diaphragm. Also attached to the diaphragm was a wire that was emersed in a metal container of acidic solution.

The user talked into the voice cone, voice sound waves caused the diaphragm to vibrate and the wire moved up and down in the acidic solution. As the wire moved up and down in the solution the resistance between the wire and the metal container changed causing the DC current to vary in proportion to the variation in sound waves.

The controversy between Bell and Gray lead to years of litigation to the level of the United States Supreme Court where a split decision gave Bell the patent for the telephone entitled Improvements in Telegraphy.

It took a little over a year for Bell to acquire and convince his wealthy father-in-law, Gardinar Hubbard, to finance the Bell Telephone Company and fund the building of the voice network infrastructure.

It's interesting to look back at the legal back and forth between Bell and Gray. It reminds me a lot of what we're seeing between Mark Zuckerberg and Facebook, the Winklevoss Twins, Wayne Chang,  Paul Ceglia.... and others.

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.

Synchronous Optical Network - SONET

Here's another entry for what I've been calling the legacy Public Switched Telephone Network (PSTN) series. In my last legacy post we covered the European or “E” carrier system. Today, let's look at SONET.

In the United States T-1 carriers have been replaced in many locations with Synchronous Optical Network (SONET) systems. Internationally, the SONET equivalent is called Synchronous Digital Hierarchy (SDH). Both SONET and SDH systems consist of rings of fiber capable of carrying very high bit rates over long distances. Copper has been replaced by fiber to inter-connect most Central Offices (CO’s) in the United States at bit rates ranging from the SONET base rate of 51.84 Mbps up to 39,813,120 Gbps. 

The base SONET standard bit rate is 51.84 Mbps and is referred to as Optical Carrier  (OC) -1 or Synchronous Transport Level  (STS) -1. SONET uses a synchronous structure for framing that allows multiplexed pieces down to individual DS-0 channels to be pulled off a SONET signal without having to demultiplex the entire SONET signal. We can look at a table of SONET bit rates.

[The OC-3072 (160 Gbps) rate level is next in the sequence but has not yet been standardized.]

The OC-1 base is used for all higher level SONET specifications. For example, a SONET specification of OC-48 can be calculated by taking the OC-1 base rate of 672 DS-0 channels and multiplying it by the OC-48 suffix of 48.

We can do the same calculation for the OC-192 specification.

It is common to run SONET rings CO to CO with all SONET connected CO’s having SONET multiplexers that can demultiplex all the way down to an individual DS-0 channel level without having to demultiplex the entire SONET frame. 

In my next legacy post I'll take a look at how SONET is used for packet-oriented data transmission (e.g. Ethernet).

Vonage and Verizon

Yesterday Vonage Holdings (2.2 million customers) was ordered by a federal jury in Alexandria to pay $58 million to Verizon Communications. The jury ruled that Vonage has infringed on Verizon patents by connecting the Vonage Internet based voice network to the Verizon Public Switched Telephone Network (PSTN).

There were actually three Verizon patents that the jury said we infringed upon by Vonage – the most significant one involves technology that links Internet voice with the PSTN. The two others involved call forwarding and wireless headsets.

In addition Verizon has asked the court to issue a permanent injunction that would stop Vonage from ever connecting to the PSTN. A permanent injunction would allow Vonage customers to only communicate with other Vonage customers using the Internet – something you can already do with many free services like Skype, Yahoo IM and Google Talk. There is a hearing scheduled for March 23 based on Verizon’s injunction request.

Last year the Supreme Court ruled to give judges in patent cases more latitude so the ruling comes as no surprise to both Vonage and Verizon. It will be interesting to see how this affects other providers offering similar domestic and long distance services.

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

Tera-bits Per Second Over Fiber

Tech.co.uk has reported that Tohoku University researchers in Japan have enabled Quadrature Amplitude Modulation (QAM) over fiber to move information at rates of hundreds of tera-bits per second. Here's a few quotes from the Tech.co.uk press release:

At the heart of the development is a technique already used in some digital TV tuners and wireless data connections called quadrature amplitude modulation (QAM). One glance at the Wikipedia explanation shows that it's no easy science, but the basics of QAM in this scenario require a stable wavelength for data transmission.

As the radio spectrum provides this, QAM-based methods work fine for some wireless protocols, however the nature of the optical spectrum means this has not been the case for fibre-optic cables ... until now.

The university team has solved the stability problem using a special laser that makes it feasible to pipe data down a glass fibre using the QAM method at blistering speeds. Although we shouldn't expect to be choosing from internet connections rated in Tbit/s anytime soon, the development could one day make us look back on ADSL as fondly as we now do our 56K modems.

Analog modems have used a form of QAM for years to move information from device to device across the Public Switched Telephone Network (PSTN) or voice network. QAM is also used by cable modems and ADSL modems to modulate (convert digital signals to analog) and demodulate (convert analog signals back to digital) communications signals.

Let's try to get a basic understanding of how QAM works - without any math! Computing devices (computers, PDA's, laptops, etc) use digital signals (1's and 0's) to process, store and manipulate information. Sending this information over long distances though typically involves a conversion or modulation of digital signals to analog signals on the sending device and a conversion or demodulation of analog signals to digital signals on the receiving device. QAM has been the method of choice for transmitting signals this way for years.

QAM combines amplitude modulation (think height of a sine wave) and phase shift (think of a sine wave moving along the x-axis relative to a zero degree reference) and allows multiple bits (combinations of binary 1's and 0's) to be transmitted for each cycle of a sine wave. I like to use the term multiple bits per cycle when I describe QAM.

QAM is categorized by the number of bits that can be transmitted in one sine wave cycle. To get a simple understanding let's take a look at 16-QAM. 16-QAM is considered rectangular QAM - the square root of 16 is 4 and this indicates that each cycle of a 16-QAM waveform can represent a 4 bit binary (1 and 0) pattern. Using the same method we can calculate 64-QAM represents an 8 bit binary (1 and 0) pattern because the square root of 64 is 8. 256-QAM can represent a 16 bit binary (1 and 0) pattern because the square root of 256 is 16, etc.

QAM signals are susceptible to instability and noise but it appears the Tohoku University researchers have figured out a way to stabilize optical signals and use QAM methods for tera-bit level data transmission. I have not been able to find any detail on the stabilization methods being used at this time.

HAPPY THANKSGIVING!

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.

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.