Data Transmission on T1 Carriers - Part 2

In Part 1of this topic I described how a T1 carrier is used to transmit data. Data transmission by nature is "bursty" meaning large amounts of information are typically transmitted and then followed by relatively quiet transmission periods. This can cause transmission problems for T-carrier systems since they rely on timing synchronization. Let's take a look how this potential problem is avoided.

Data Transmission on T-1 Carriers Part 1

Back in December I wrote a post here titled T1 Lines - What They Are. In the post I discuss the Digital Signal (DS) Level System and how combining the equivalent of 24 DS-0 voice channels along with overhead consisting of timing and synchronization bits brings the DS-1 bit rate to 1.644 Mbps - that's a T1. In this post, let's have a look in more detail to get a better idea of how the entire system works. 

The T-1 Carrier uses time division multiplexing and was designed for voice call transmission. When used for data one would think it would be possible to achieve a data bit rate of 64 Kbps over a T-1 carrier. Looking a little closer one sees that data on T-1 carriers is transmitted in the form of only 7 bit words, all eight bits are not used. Why? 

Remember the T carrier system was initially designed for voice. The first signal synchronization used for the T-1 carrier substituted a single in band signaling bit, used for control, for each of the 24 channels in every sixth frame. This means in the sixth and twelfth frames of every T-1 carrier master frame there is a bit used for in-band signaling. This is referred to as bit-robbing. Bit robbing is usually not a problem when transmitting voice. Even though the signal is slightly distorted, the listener on the receiving end cannot perceive the distortion. However this is a major problem when transmitting data as any data received with missing bits will be distorted and received incorrectly. To eliminate the problem caused by bit robbing data on the T-1 carrier is limited to seven bits per frame in all frames. By decreasing the number of bits transmitted the data bit rate is reduced.

For this reason, 56 Kbps Clear Channel Capability is the term used to refer to the T-1 carrier single channel maximum data bit rate.

T-1 Carrier Pulse Cycles

If we look closer at a T-1 Carrier signal we see there are negative and positive pulses combined in the digital pulse train. A sample T-1 signal pulse train is shown in the figure below.

Sample T-1 Pulse Train

It has been found that alternating positive/negative pulse trains (bipolar) produces fewer transmission errors than all positive or all negative pulse trains. These pulses are used to represent binary 1’s and each pulse, when non-zero, is positive half the non-zero cycle (50%) and negative half the non-zero cycle. We can look at an example of a positive (cycle 1) and negative (cycle 4) pulse from the above figure.

Sample T-1 Positive and Negative Going Pulses

In the figure above, T represents the period, or time it takes to complete a single pulse cycle. We can calculate the percent duty cycle using the following equation:

The pulses here are not zero for one half of the pulse period and have a 50% duty cycle. Let’s go back now and look at the original pulse train diagram and look at each cycle:

You can now see that if a pulse is present within a cycle time slot, whether positive or negative, it represents a 1 bit and if no pulse is present, it represents a 0-bit.

In Part 2 of this series I'll cover something called Bipolar with Zero Substitution (B8ZS) for T-1 signal synchronization.

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).

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.

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.

What's a T3 Line?

In my last post I described what a T1, also called a DS-1, line was. Most of us have also heard the "T3 Line" term used. Let's take a look at what a T3 or DS-3 line is.

DS-2 Signal
Before we can describe a DS-3 line, let's first take a look at a DS-2. In that last post we figured out how each DS-1 signal (T1 line or circuit) carries a bit rate of 1.544 Mbps. Four 1.544 Mbps digital DS-1 signals are multiplexed into one DS-2 signal. If we have 4 DS-1 signals per DS-2 signal and each DS-1 signal is 1.544 Mbps we can calculate:

Adding overhead consisting of timing and synchronization bits brings the DS-2 bit rate to 6.312 Mbps.

DS-2 Formation

DS-3 Signal

Each DS-2 signal carries a bit rate of 6.312 Mbps. Seven 6.312 Mbps digital DS-2 signals are multiplexed into one DS-3 signal. If we have 7 DS-2 signals per DS-3 signal and each DS-2 signal is 6.312 Mbps we can calculate:

Adding overhead consisting of timing and synchronization bits brings the DS-3 bit rate to 44.736 Mbps.

And..... 44.736 Mbps.... that's a T-3 line!

T1 Lines - What They Are

Most of us have heard about "T1" lines. We know they are some kind of (expensive) communications line you can get from one of the telephone companies. It turns out T1's are part of the Digital Signal (DS) Level System. 

Back in August, I wrote a post titled More on CODECs: Quantization + Sampling Rate = A PCM Wave. In that post I described how a piece of an analog signal is quantized and companded and then given an 8 bit binary code in a process referred to as encoding. From that post, we know to convert an analog signal to a digital signal the analog signal is sampled 8000 times per second and, after matching the instantaneous voltage sample level to one of 256 discrete levels, an 8 bit code is generated for each sample. If we multiply the sample rate by the bit code we get:

(8000 samples/second)(8 bits/sample) = 64,000 bits per second (bps)

So we can say a single analog voice channel, after conversion from analog to digital, requires 64Kbps of digital bandwidth. This 64Kbps is referred to as Digital Signal Level 0 (DS-0) and is the basic building block or channel for the existing digitally multiplexed T carrier system in the United States and the digital E carrier system used in Europe. 

Voice calls are digitally multiplexed using either time division multiplexing or statistical time division multiplexing. Calls are grouped in a way similar to frequency division multiplexing. Let’s look at how this is done.

Digroups or DS-1 signals

Individual analog voice call channels converted to digital and require a bit rate of 64 Kbps each. 24 64 Kbps digital voice channels are multiplexed into digroups or DS-1 signals. If we have 24 DS-0 signals per DS-1 signal and each channel is 64 Kbps we can calculate:

Adding overhead consisting of timing and synchronization bits brings the DS-1 bit rate to 1.544 Mbps - that's a T1!

DS-1 Formation

DS-1 Overhead

We’ve described the process of encoding where an analog signal is sampled 8000 times per second, quantized into one of 256 discrete signal levels, companded it is then given an 8 bit binary code. After a single analog signal sample has been encoded it is multiplexed, with 24 other encoded 8 bit sample signals. This generates a 192 bit (8 bits/sample signal × 24 sample signals) sequence for the 24 sample signals. A process called framing then adds one framing bit to create a 193 bit frame.

DS-1 With Overhead

The framing bits are used to keep the receiving device in synch with the frames it is receiving. Every twelve frames are grouped into a masterframe, also referred to as a superframe. Included within each masterframe is a twelve bit frame pattern from the 12 grouped 193 bit frames This twelve bit frame pattern carries a bit pattern of 000110111001 and repeats itself with each masterframe.

Masterframe

This masterframe bit pattern is used for synchronization.

Remember each channel is sampled 8000 times per second so a single frame represents one eight-thousandth of 24 individual channels or telephone calls. We can also say that, in one second a DS-1 signal transmits 8000 193 bit frames. We can use these numbers to calculate the true DS-1 bit rate which includes both data and overhead (framing) bits:

Each DS-1 signal carries a bit rate of 1.544 Mbps and.... that's a T1!

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.

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.