No T1 Lines in Europe - The E-Carrier Hierarchy

Today I'll continue with a post on what I've been calling the legacy Public Switched Telephone Network (PSTN). In my last legacy post we covered T-4 and T-5 lines, today let's take a look at the European or “E” carrier system.

The European or “E” digital transmission format is slightly different than the North American T-carrier system format. With the E-Carrier system we are still taking individual voice call analog signals and converting to a digital signal by sampling the analog signal 8000 times per second and, after matching the instantaneous voltage sample level to one of 256 discrete levels, generating an 8 bit code for each sample. We are still dealing with the fundamental DS-0 building block of 64Kbps of digital bandwidth per single analog voice channel we used for the T-Carrier system. The differences between E-Carrier and T-Carrier deals with the number of channels and how these channels are used. Let’s start by looking at a European E-1 system and how it compares to a North American T-1 system.

The E-Carrier system starts by multiplexing 32 DS-0 channels together to form an E-1 circuit while the North American T-Carrier system multiplexes 24 DS-0 channels to form a T-1 circuit. 

The 32 DS-0 channels of an E-1 circuit combine from Channel 0 up to Channel 31. Channel 0 is used for framing (synchronization), channels 1-15 and 17-31 are used for individual DS-0 channels and Channel 16 is reserved and not used.

This system is also referred to as the “30 plus 2 system” because an E-1 signal consists of 30 DS-0 signals used for voice plus Channel 0, which is used for overhead and Channel 16 which is not used at all. In the European system, all synchronization (framing) is handled by Channel 0 so framing bits are not required on individual DS-0 channels.

We can calculate the signal rate for an E-1 circuit as follows:

E-2 through E-5 are carriers in increasing multiples of the E-1 format. We can look at a table showing DS data rates and how they correspond to the European E Carrier system.

In my next legacy PSTN post I'll cover the Synchronous Digital Hierarchy (SDH) system.

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).

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.

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).

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.

DS-4 and DS-5 Lines

It's been a while since I've posted on what I've been calling the Legacy Public Switched Telephone Network (PSTN). My last related post was way back on December 15, 2011 titled What's a T3 Line? Today, Let's take a look at higher bit rate signals in the DS system.

DS-4 Signal
Back on December 15th, we said each DS-3 signal carries a bit rate of 44.736 Mbps. Six 44.736 Mbps digital DS-3 signals are multiplexed into one DS-4 signal. If we have six DS-3 signals per DS-4 signal and each DS-3 signal is 44.736 Mbps we can calculate:

Adding overhead consisting of timing and synchronization bits brings the DS-4 bit rate to 274.176 Mbps.

DS-4 Formation

DS-5 Signal

Each DS-4 signal carries a bit rate of 274.176 Mbps. Two 274.176 Mbps digital DS-4 signals are multiplexed into one DS-5 signal. If we have two DS-4 signals per DS-5 signal and each DS-4 signal is 274.176 Mbps we can calculate:

Adding overhead consisting of timing and synchronization bits brings the DS-5 bit rate to 560.16 Mbps.

DS-5 Formation

One DS-5 channel can carry 8064 voice channels.

We can look at a table showing these DS data rates and how they correspond to the North American T Carrier system.

Looking at the table it is easy to see that the DS-0 signal level is the foundation for the entire T Carrier hierarchy in North America. Notice one DS-1 line is the equivalent of 24 DS0 64 Kbps DS-0 voice channels. Also notice that one DS-2 line is the equivalent of 4 DS-1 lines or 96 DS-0 voice channels.

Copper wire pairs can be used to transmit at levels up to DS-2. At levels above DS-2 coaxial cable, fiber or microwaves must be used.

In my next Legacy PSTN post I'll cover the European (E) Carrier System.

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 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.

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

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.

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.

SONET Packet-Oriented Data Framing

In my last legacy PSTN post I discussed how Synchronous Optical Network (SONET) is used to multiplex, transmit and then de-multiplex voice calls. Today, let’s take a look at how SONET  is being used to transmit packet-oriented data (in today’s world - basically Ethernet).

In that last SONET post we said the SONET international equivalent is called Synchronous Digital Hierarchy (SDH). Now, when we talk about data at the SONET/SDH level we’re talking frames (think layer 2 OSI model) and the base unit of framing for SDH is something called a Synchronous Transport Module, level 1 (STM-1) with operates at 155.52 Mbps. 

In the post I also said 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. Now, because we’re talking 3 times an STS-1 and it is concatenated (combined), the base SONET data framing unit (running at 155.52 Mbps)  is referred to as a STS-3c (Synchronous Transport Signal 3, concatenated) which is also referred to as an OC-3c (Optical Carrier - 3c). 

Now that I have you completely confused (!) lets’s talk a little more about packet frames. A typical packet frame consists of a header, payload (the actual data being sent) and some kind of trailer. I like to use a letter analogy to understand what is going on - someone writes a letter (think of the letter as the payload or data). It gets put on an envelope (think of the envelop as the header and trailer for now). At the sending end the letter gets a destination address, a return address, etc and gets delivered. At the receiving end the letter gets opened, the envelop discarded and the letter itself saved and used.

For an STS-3c framing unit, the payload rate is 149.76 Mbit/s and overhead is 5.76 Mbit/s.

If we look at an individual SONET STS-3c frame - it’s  2,430 octets long. SONET systems transmit nine octets of overhead and then 261 octets of payload in sequence. This transmission is  repeated nine times in 125 micro-seconds until 2,430 octets have been transmitted. 

Timing is critical here (that's why it's called synchronous) for communications across the entire network.

87 Million 4G Devices Will Ship In 2012

According to a recent ABI Research report, 4G devices are moving rapidly from the assembly line to retail stores.

Here's some details:

• Refers to a range of 4G-enabled mobile devices, from USB dongles, smartphones, tablets, 4G portable hotspots, and wireless broadband CPE modems
•  4G devices are expected to generate 87 million in unit sales in 2012, up 294% year-on-year.
• 61 million 4G handsets being shipped in 2012.
• 26 million 4G non-handset products (e.g. USB dongles for legacy laptops and netbooks, by premise equipment, home modems, etc) will be shipped
• The lion’s share of the market is now backing LTE as service provider and vendor support has fallen away from WiMAX.
• There is a natural evolutionary demand from 3G end-users, both business and consumer, to jump onto the 4G data bandwagon. 
• Mobile device vendors are experiencing intense competitive pressure, which is expected to bring down LTE handset prices, estimated at 10 to 20 percent over the next two years.

It's not all good though. There are still some big technical issues that need to be worked out including the recent Australian iPad 3 promotion fiasco, when iPad 3s were being promoted as being ‘LTE-ready,’ even though the modem is unable to access the Australian LTE spectrum band. 

In addition, some customers will not be ready this year to pay a premium for 4G handsets and 4G services. 

As a reference, in a February 2012 report Forrester predicted by 2016 one billion people will own smart phones. 

U.S. Needs More Cyber Security Training and Education

Richard Marshall, director of global cyber-security management at the Department of Homeland Security made some interesting comments yesterday at the FOSE government IT show in Washington, DC. FOSE is a conference focused on cyber-security issues facing the public sector and what it means for protection against threats, cloud computing and new open government directives.

Here's a few quotes Marshall made at the conference taken from a post over at esecurityplanet.com

Working in concert with the government, the private sector has made significant strides in improving software security and ferreting out vulnerabilities in the supply chain, but the flow of cyber-security experts graduating from the nation's universities with advanced degrees remains anemic.

One of the most important steps policymakers can take is to nourish the education and training of a new crop of security expert.

No matter how successful we are in those two elements, we are going to fail if we don't invest more money, time, attention and rewards to educate the workforce. That's our legacy-to-be.

"The IT industry provides a one trillion -- with a 'T' -- dollar contribution to the U.S. gross domestic product. If you're looking for a metric for cyber-security, money is a good metric.

And my favorite quote from the piece which I'll probably catch some flack for posting:

Look at all the great football and basketball programs. They're all on scholarships. They're not playing for fun -- they're playing for money. We need to do the same thing with our computer science students.

Nicely said.

Blog in Transition

As some of you know I've gone through a few career-related changes over the past year. After seventeen years the ICT Center was sunset by the National Science Foundation. Seventeen years is a long time for anything to be funded by the NSF and I'd like to thank everyone involved - there are so many - from all over the United States. It was so much fun to do the work we did at the historical time we did it. We positively influenced tens-of-thousands of lives in our country - our legacy - and that is pretty cool.

On Sept 5, after a lot of contemplation, I early-retired from a tenured faculty position (basically a job for life) in Massachusetts and with that, many are convinced I have truly lost my mind :) 

Seriously, the time seemed right for me to do something a little different while I was still young and nimble enough! The opportunity was there and I accepted a four-day-a-week position as an Associate Director of the National Center for Optics and Photonics Education (www.op-tec.org).  OP-TEC is another NSF Advanced Technological Education center of excellence located in Waco, TX and funded via the University of Central Florida. I get to Waco once a month for four days. I'm also doing a little bit of evaluation work with NSF funded projects around the country

Hmmmmm, optics and photonics you may ask? "What the heck is that? I thought you were some kind of communications guy??" Well...... when I first started teaching (30 years ago) I taught both geometric optics and wave optics for a number of years. In fact, the faculty position I just retired from included seniority in the laser electro-optics department. It's been great to dig in to some familiar content and dust off my old notes again. I've also had to dust off some brain cells and that is always a good thing. I'm seeing lots of intersections and opportunities with optics, lasers and computers. Hmmmmm

So..... what's up with this blog? I'm feeling settled in now and am ready to start writing again. I'll be writing about many of the same emerging technology topics as the past and a bunch of new topics. I'm expanding horizons, learning new stuff, making mistakes as I learn, and learning from those mistakes. Not ready to spend my time fishing and golfing - I'm actually pretty good at fishing but really stink at golf :) At least not yet. 

Thanks to everyone who has followed me here in the past. You thought you got rid of me but - I'm back :)

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 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.

Distributed Inference And Tesla With Some SETI Nostalgia

In this post, I’m setting aside any political stuff and focusing solely on tech.

 

In recent months, the electric vehicle (EV) market has seen a decline, marked by falling sales and an increase in unsold inventory. Tesla, in particular, has received a significant share of negative attention. During Tesla's first-quarter earnings call last week, Elon Musk diverged from the norm by highlighting Tesla's broader identity beyond its role in the automotive industry. He emphasized the company's engagement in artificial intelligence and robotics, suggesting that pigeonholing Tesla solely within the EV sector overlooks its broader potential.

 

Musk's suggestion to actively utilize Tesla's computational power hints at a larger strategic vision. He envisions a future where idle Tesla vehicles contribute to a distributed network for AI model processing, termed distributed inference. This concept could leverage the collective computational strength of millions of Tesla cars worldwide, extending the company's impact beyond transportation.

 

Very interesting – I drive maybe 1-2 hours per day, the rest of the time my car is not being used. What if all that computing horsepower could be used while I’m not using it? Musk’s concept brings up memories of the sunsetted SETI@home computer application. SETI was a distributed computing project that allowed volunteers to contribute their idle computer processing power to analyze radio signals from space in the search for extraterrestrial intelligence (SETI). SETI@home used data collected by the Arecibo Observatory in Puerto Rico and the Green Bank Telescope in West Virginia to search for patterns or anomalies that could indicate the presence of intelligent alien civilizations.

 

Participants in SETI@home downloaded a screensaver or software client onto their computers, which would then process small segments of radio telescope data during periods of inactivity. The processed data would be sent back to the project's servers for analysis. By harnessing the collective power of millions of volunteer computers around the world, SETI@home was able to perform computations on an unprecedented scale. The project was launched in 1999 by the University of California, Berkeley, and it quickly became one of the largest distributed computing projects in history. Although the original SETI@home project ended in 2020, its legacy lives on as an example of the power of distributed computing and the widespread public interest in the search for extraterrestrial life.

 

Musk's vision underscores Tesla's potential to revolutionize not only the automotive sector but also broader domains such as artificial intelligence and robotics. It signifies a strategic shift towards leveraging Tesla's resources and expertise in a SETI-like way to drive innovation and create value in new and unexpected ways.