We are experiencing a breakneck growth in the interconnection of personal computers, terminals and telephones in the business environment. T1 technology is proving to be a cost-effective means of linking voice and data, both inter-office and intra-office, and serves as an alternative to high speed modems for data transport. There is significant discussion these days about "T1 Gateways" and "T1 trunks" as the cost from the various phone companies of these services goes down. Users are discovering that it costs less to have a T1 trunk than a series of leased telephone lines in a point-to-point topology. This increase in the use of T1 requires a fundamental understanding of the technology.
T1 is a high speed digital network (1.544 mbps) developed by AT&T in 1957 and implemented in the early 1960's to support long-haul pulse-code modulation (PCM) voice transmission. The primary innovation of T1 was to introduce "digitized" voice and to create a network fully capable of digitally representing what was up until then, a fully analog telephone system.
Perhaps the way to really begin this discussion is to discuss the AT&T Digital Carrier System referred to as "ACCUNET T1.5". It is described as a "two-point, dedicated, high capacity, digital service provided on terrestrial digital facilities capable of transmitting 1.544 Mb/s. The interface to the customer can be either a T1 carrier or a higher order multiplexed facility such as those used to provide access from (fiber optic) and radio systems."
So in the basic definition there is the discussion that there is a "higher order" or hierarchy of T1. There is T1 which is, as we have discussed, a network that has a speed of 1.544 Mbps and was designed for voice circuits or "channels" (24 per each T1 line or "trunk"). In addition, there is T1-C which operates at 3.152 Mbps. There is also T-2, operating at 6.312 Mbps, which was implemented in the early 1970's to carry one Picture phone channel or 96 voice channels.
There is T-3, operating at 44.736 Mbps and T-4, operating at 274.176 Mbps. These are known as "supergroups" and their operating speeds are generally referred to as 45 Mbps and 274 Mbps respectively.
Analog Transmission
The telephone system that evolved from the days of Alexander G. Bell was designed around providing analog dialup telephone service. Everything was based on voice communication services on a switched (nondedicated) basis. A user could use his or her telephone to connect with another user on the network either through the operator-connected or dialup-addressing scheme.
Since voice was the primary service provided, the telephone set evolved into a device that took the sound wave from the human vocal cords and converted that sound into an electrical current that was represented by its analog equivalent. The human voice produces constantly changing variables of both amplitude (the height of the wave) and frequency (the number of cycle changes per second).
As these constantly changing variables of amplitude and frequency are produced, an electrical wave is produced. The telephone set converted the sound wave into electrical energy that will be carried down a pair of wires. As the electrical energy was introduced to the wires, certain characteristics began to work on the energy. (Resistance, Noise, Crosstalk, Electricity from other conversations.) This was an undesirable way of getting calls from one place to another.
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Number of Cycle Changes in 1sec
Tl Fundamentals
The Evolution to Digital: The T-Carrier System
Thus, the T-carrier system was born. The telephone companies wanted to enhance the quality of calls and better use the cable facilities. The evolution to T-carrier was important for a number of reasons:
1. It was the first successful system designed to use digitized voice transmission.
2. It identified many of the standards used today for digital switching and digital transmission, including a modulation technique.
3. The transmission rate was established at 1.544 Mbps (1,544,000 bits per second), which became the building block for the North American digital hierarchy and the AT&T digital standard.
4. The original application of the T-carrier technology defined many of the rules, or protocols, and constraints in use today for other types of communication. Remember that the carrier system evolved, to service voice dialup communications.
The telephone companies saw this as a telephone company service only.
As more central offices were added to the network, the need for wires grew.
These techniques were first employed to provide lower cost, better quality dial-up telephone services. This same technology is the underling idea of full end-to-end digital networking services, that is the basis of the Integrated Services Digital Networks (ISDN).
Analog to Digital Conversion
Digital architecture, dictates the use of a digital bit stream, the analog wave must be converted into a usable format; that is digital pulses represented by a 0 or a 1.
As the analog signal is transmitted to the network, it must be
converted from varying amplitudes and frequencies to a digital format,
that digital format is represented in the form of Is and Os
(or presence and absence of a voltage).
The analog wave must be sampled enough times and converted to create a stream of Is and Os that is precise enough to be recreated at the distant end, producing what sounds like the original conversation. According to the Nyquist rule, a digital signal should be created by sampling the analog wave at twice the highest frequency on the line. This frequency on the line is represented in hertz.
The telephone company delivered 4,000 Hz or (4 kHz) line capacity. Using the Nyquist rule (communications theory) the sampling of an analog wave at twice the maximum frequency on the line meant that 8,000 samples per second should be sufficient to re-create the wave for the digital-to-analog and analog-to-digital conversion.
This yields: 4,000 cycle changes per second (4kHz) line capacity
x 2 sampling rate of the line for accuracy 8,000 samples per second required
The sample now had to be created into a bit stream of Is and Os. Enough bits must be used to create a digital word. Using 8-bits, represented enough quality to create this word. Using an 8-bit word creates enough different points on the wave to adjust for the fluctuation of the human voice. The wave is divided into 256 possible points along the amplitude at the moment that the sample is taken.

Amplitude when sample is taken
Using two states (1 or 0) and 8 bits, this is created in binary math to be 28 or 256 points on the analog wave. The 1 represents the presence of voltage, and the 0 represents the absence of voltage. This conversion uses PCM (Pulse Code Modulation), a technique to create the sample. Using PCM and the rules established, the transmission of digital equivalent of the analog wave results in: 8,000 samples/second x 8 bits per sample = 64,000 bits per second.
T-l is:
1. A four wire circuit. Since this technology evolved from the old twisted-pair environment, four wires were used. This gives us two wires for transmit and two wires for receive. (Pins 1,2 and 4,5)
2. Full duplex. Transmission and reception of data can take place simultaneously.
3. Digital. This is an all-digital service. Data, analog fax and analog voice are all converted into digital pulses (Is and Os) for transmission on the line.
4. Time-Division Multiplexing. The digital stream is capable of carrying of 64-kbps channel; 24 channels are multiplexed together to create and aggregate of 1.536 Mbps. Time-division allows a channel to use a slot one-twenty-fourth of the time. These can be fixed timeslots made available to the channel.
5. Pulse code Modulation. The Analog voice is sampled 8,000 times per second; an 8-bit word represents each sample, thus yielding the 64-kbps(kilobits per second) channel capacity.
6. Framed Format. As the pulse code modulation scheme is used, the 24 channels are time-division multiplexed into a frame to be carried along the line. Each frame represents an 8-bit sample from each of the 24 channels. Added to this is a framing bit. The net result is a 193-bit frame. There are 8,000 frames per second, therefore a frame is 125 microseconds long. Framing accounts for 8-kbps overhead (1 bit x 8,000 frames). Adding this 8-kbps to the 1.536 Mbps you get 1.544 Mbps.(MegaBits per second).
7. Bipolar Format. Remember Tl uses electrical voltages on the line to represent the pulses (Is). The bipolar format serves two purposes: it reduces the required bandwidth (which increases repeater spacing), and it averages out the signal voltage to zero to allow dc power to be simplexed on the line to power intermediate regenerators. Every other pulse will be represented by the negative equivalent of the pulse. So, the first pulse will be represented as a positive +3 volts, the next pulse will be represented by a negative -3 volts. This bipolar format is also called alternate mark inversion (AMI).
8. Byte Synchronous. Timing happens on the channels from the pulses that appear within the sample. This timing keeps everything in sequence. If the devices at both ends of the line do not see any pulses, they would lose track of where they were. This is also known as the "one's density rule", in every 24 bits of information to be transmitted, there must be at lease three pulses, and no more than 15 zero's may be transmitted consecutively. AT&T set down a more stringent rule that states: "In every 8-bits of information, at least one pulse must be present".
1.3 Frame Format Go to Top
It looks like this: Original Data
0000000 0 0000000 0 Data- after pulse stuffing
0000000 1 0000000 1
This works fine for voice transmission, however for data transmission this poised a problem. This limited data transmission to only 56K. A technique was developed to overcome this pulse stuffing idea and still meet the "one's density rule".
Its known as bipolar 8 zero substitution (B8ZS), this is implemented in the CSU. The CSU reads the 8-bit format, recognizing that a string of 8 zeros will cause problems, it strips off the 8-bit byte and substitutes a "fictitious byte". Which looks like "00011011". The receiving CSU sees this "fictitious word" strips it off and sends all zeros for delivery. Thus the ability to transmit all usable data at 64-kbps "clear channel" becomes a reality.
The digital hierarchy of North America includes the following: DSO Twenty-four DSO's combined produce the DS1.
DS1 The digital signal 1 is a TDM-PCM aggregate of 1.544 Mbps, regardless of the medium used to carry the signal. Referring to a Tl as a DS 1 is often heard in the industry. To clarify this statement, Tl is the first level of the T-Carrier system. DS1 is the multiplexed digital signal, first level, carried inside the T-Carrier.
DS1C
DS2
DS3
DS4/NA
DS4Two DSI's
Four DS 1 's multiplexed together
28DSl's
Three DS3's
SixDS3's
ANSI Tl.107 Rates
Designator
Capacity
Equiv
DS1
DSO
64
-
1
DS1
1.544
1
24
DS1C
3,152
2
48
DS2
6,312
4
96
DS3
44,736
28
672
DS4/NA
139,264
84
2,016
DS4
274,176
168
4,032
Frame Synchronization:
The frame is a group a data bits in a specific format, with a flag at each end to indicate the beginning and end of the frame. You no it as D4 superframe, with D4 the framing pattern is a 12-bit sequence, which allows "robbing" of the signaling bits in the 6th and the 12th frames of the superframe. The 8th bit from each sample in frames 6 and 12 provides signaling. Thus in-band signaling, which makes each channel 56-kbps of useable space.
Signaling
Remember the need to find bits earlier where in the 6th and the 12th frames we sent signaling on the 8th bit. Well, bit robbing, or stealing the 8th bit in each of the channels
(1-24) in these two frames allows enough bits to signal between the transmit and receive ends. The most common form of signaling on a Tl is 4 wire E&M type 1,11,111. The most commonly used however is type 1.
Signaling is used to the receiver where the call is destined to end up. Some common types of signals include:
On Hook
Dial Tone
Dialed Digits
Ringing Cycle
Busy Tone
Clocking Go to Top
What time is it? No not really, but timing is very important on a digital signal. Remember we have these bits of information that are being transmitted onto the line, well if we transmit them faster than the receiving end can except them this could result in lost bits. So we want to send them at the same rate the other end can receive them.
Using the PCM technique, 8-bits are encoded from each sample and placed into a timeslot. Aha "timeslot". The same principal as stated in the previous paragraph applies. As signals and links are processed through the network, it is the 8-bit pattern that is routed from time slot to time slot.
The Master-Slave technique is the easiest form of clocking, this means each node on the network takes its timing from the incoming stream of data.
Using an outside clock source or timing for all nodes on a network allows a higher quality clock source.
Extended Superframe
Performance Issues
The D4 superframe allows signaling (robbed bit) and ones density (stuffed bits). These two capabilities, combined with the framing bits, were constantly taking channel capacity from the Tl. Thus giving us a 56-kbps data stream. The robbed or stuffed bits were designed around control, yet with the need for maintenance and diagnostic capabilities, no spare capacity existed.
In the early 1980s, AT&T suggested the implementation of ESF to provide non-disruptive error detection and problem non-service-affecting diagnostics on Tl circuits. What a novel idea.
What Does ESF Do?
Extended superframe format, as implied by the name, extends the superframe from 12 consecutive frames of information to 24 frames of information. The 8-kbps(kilobits per second) overhead originally used strictly for framing is now subdivided into three functions.
Original SF (or D4) 8-kbps for framing only
ESF (Extended Superframe)
2 kbps for framing 2 kbps for error detection 4 kbps for facility data link
The facility data link is a synchronous communications channel. These are all CSU messages, that are sent back and forth across the network. Go to Top
A digital source, or terminal, is the equipment that generates digital signals for transmission through the digital network. The large majority of digital sources now produce a DS-1 signal. The D4 Channel Bank is an example, although it can produce signals at other rates as well.
The reference to the term "Channel Bank" is made quite often in the T-1 language. The type of Channel Bank is important since it defines the type of formatting that is required. For example, a D4 Channel Bank must have a DS-1 signal with data formatted in accordance with the D4 format.
The purpose of a Channel Bank in the telephone company is to form the foundation of multiplexing and demultiplexing the 24 voice channels (DS0). The D-type Channel Bank is used for digital signals. There are five kinds of Channel Banks that are used in the System: D1, D2, D3, D4, and DCT (Digital Carrier Trunk).
A transmitting portion of a Channel Bank digitally encodes the 24 analog channels, adds signaling information into each channel, and multiplexes the digital stream onto the transmission medium. The receiving portion reverses the process. As these were designed as voice circuits, the assumption is that the digital data is PCM voice and that the voice is companded and expanded through the use of CODECs. D1 banks (later called D1A) were first installed in 1962 and their success led to modifications of D1B and D1C. The original D1A,B, and C banks used 7 bits for each voice sample and one bit in each code word for carrying the signaling (off hook, ring, etc). When it became desirable to connect several T1 transmission spans together, the performance was not too good. In addition, it was realized that providing signaling information in every code word was wasteful since 8,000 bits per second was not required to provide the signaling information for a channel; the signaling information simply did not change that quickly.
As a result of these conditions, another modification to the D1 series (D1D) and the new D2 channel bank were developed. The D2 bank uses all eight bits of every time slot to encode the analog signal except for selected frames. Supervisory and signaling information is sent by using the least significant bit from the code word in each channel every sixth frame. The companding characteristic also was changed to give better performance. The D2 bank increased the packing density to 96 channels in the same space as the 72 channels for a D1 bank.
D3 and D4 banks were motivated by advances in ICs, allowing packaging of 144 channels in a single bay. Following the D4 bank, advances in technology resulted in the development of the Digital Carrier Trunk unit, or DCT. It was developed by the Bell System to be smaller, lower cost, and easier to maintain than the D4 channel bank.
The D1 type channel bank (D1A,B,C) placed alternate 1's and 0's in the 193rd bit position. It was assumed that random data would not contain this pattern, in bits spaced exactly 193 bits apart, for any significant length of time. The receiving device would find the 193rd bit by using a simple search technique. This algorithm had the advantages of circuit simplicity and speed. In the early 1960's, there were few commercially available ICs for building complex logic functions, and elementary designs cost less. The disadvantages of this technique were rapidly uncovered when equipment was installed in actual customer sites. Certain standard analog tones, such as the 1000 Hz test tone, applied to one or more voice channels and digitized by Channel Bank, created an alternating one and zero pattern every 193 bits in one or more voice channels. It was possible for the terminal to lock up on the incorrect pattern. This condition, affecting all 24 channels, could last until the test tone was removed. The 1000 Hz tone has been changed to a 1004 Hz test tone.
By the time this problem became apparent, it had been decided to use T-carrier for toll quality telephony, which required more precise coding techniques. D1 channel banks used seven bit encoding for voice signals, and an eighth bit for signaling. The new format provided for eight bit coding most of the time (5/6 frames) and seven bits only in one frame out of six. This is known as 7 5/6 coding with "robbed bit" signaling and was first implemented in the D2 channel bank (D1D is a retrofit of D1 channel banks with D2 capability).
Besides the "false frame" problem, D2 bank designers were faced with a new set of problems. The new format required two steps; first, find the 193rd bit, and second, find the sixth and 12th frame in a 12-frame sequence. The time required to find the proper bit sequence rises exponentially as the number of bit positions between frame bits increases. Although we still use every 193rd bit, it is time-shared between the terminal framing pattern (odd numbered frame bits) and the superframe alignment pattern (even numbered frame bits). Finding the 193rd bit position was still based on an alternating 1's and 0's pattern, but now it only appeared in every other 193rd bit.
The new technique provided for increased "false frame" protection.
The downside of the technique was that the time to reframe was much longer. With
the D2 format the maximum average reframe time (MART) would be about 200
milliseconds. This was too much time to be out of service so new algorithms were
developed that decreased the time to 50 msec which is now the specification
standard. Succeeding channel bank equipment (D3 and D4) used the same framing
sequence as D2. In fact, the Superframe Format is most often referred to as the
D4 frame format even though it began with D2. This sequence defines a
"superframe" consisting of two interleaved patterns. The terminal
framing pattern ("F" bit) is a repeating ones and zeros in odd
numbered frames and the superframe alignment pattern ("S" bit) is
"001110" in the even numbered frames. This results in a 12-bit
superframe pattern of:
|
Odd Six Bits |
Even Six Bits |
Combined Twelve Bits |
|---|---|---|
|
101010 |
001110 |
100011011100 |
The D4 Format is shown in Figure 4 below. Notice that the "F" bit and the "S" bit are all called "S bits". While this is confusing, it is a terminology remnant of the time when there were only "S" bits (vis-à-vis D1 format).
|
Frame # |
S-bit terminal Framing (Ft) |
S-bit signal Framing (Fs) |
Information bits |
Signaling bit |
Signaling Channel |
|---|---|---|---|---|---|
|
1 |
1 |
- |
1-8 |
- |
|
|
2 |
- |
0 |
1-8 |
- |
|
|
3 |
0 |
- |
1-8 |
- |
|
|
4 |
- |
0 |
1-8 |
- |
|
|
5 |
1 |
- |
1-8 |
- |
|
|
6 |
- |
1 |
1-7 |
8 |
A |
|
7 |
0 |
- |
1-8 |
- |
|
|
8 |
- |
1 |
1-8 |
- |
|
|
9 |
1 |
- |
1-8 |
- |
|
|
10 |
- |
1 |
1-8 |
- |
|
|
11 |
0 |
- |
1-8 |
- |
|
|
12 |
- |
0 |
1-7 |
8 |
B |
Figure 4 - The D4 Format
As early as 1979, AT&T proposed the Extended Superframe Format be
implemented on its T1 circuits in order to provide in-service diagnostic
capabilities as well as improved false frame protection. With ESF, the 193rd bit
is now time shared by three functions: frame synchronization bits; CRC-6 bits;
and Facility Data Link (FDL) bits. Frame synchronization bits are carried in six
of the 24 bit positions provided by the 193rd bit. These are in the 4th, 8th,
12th, 16th, 20th, and 24th positions and the pattern is "001011". This
simple six-bit pattern performs both the "F bit" and "S bit"
functions of the D4 superframe. "False frame" sensitivity is
eliminated by using the CRC-6 error checking bits to determine which of several
"candidates" for the frame bit are the actual 193rd bit. CRC-6 uses a
mathematical algorithm to check the contents of the entire superframe (all 4632
bits) and obtains a 6-bit (hence its name) coded "signature" for those
data bits. The FDL may be used for any purpose, but is ideally suited for
communicating ESF performance information from local, remote, and intermediate
equipment along a facility and for sending control commands for protection
switching, network and remote equipment configuration, etc. In essence it is a 4
Kbps channel embedded in the T1 format. Bellcore document TR-TSY-000194
(Extended Superframe Format Interface Specification - December 1987), ANSI
T1.403-1989, and AT&T Publication 54016 describes how this channel may be
used. This includes the format of the messages , commands, and responses. Most
CSU's today interpret these commands and execute the appropriate responses. The
ESF Format is shown is Figure 5.
|
Frame # |
Fe bit |
DL bit |
CRC-6 |
Info bits |
Signaling bit |
Signaling channel |
|---|---|---|---|---|---|---|
|
1 |
- |
m |
|
1-8 |
- |
|
|
2 |
- |
- |
C1 |
1-8 |
- |
|
|
3 |
- |
m |
|
1-8 |
- |
|
|
4 |
0 |
- |
|
1-8 |
- |
|
|
5 |
- |
m |
|
1-8 |
- |
|
|
6 |
- |
- |
C2 |
1-7 |
8 |
A |
|
7 |
- |
m |
|
1-8 |
- |
|
|
8 |
0 |
- |
|
1-8 |
- |
|
|
9 |
- |
m |
|
1-8 |
- |
|
|
10 |
- |
- |
C3 |
1-8 |
- |
|
|
11 |
- |
m |
|
1-8 |
- |
|
|
12 |
1 |
- |
|
1-7 |
8 |
B |
|
13 |
- |
m |
|
1-8 |
- |
|
|
14 |
- |
- |
C4 |
1-8 |
- |
|
|
15 |
- |
m |
|
1-8 |
- |
|
|
16 |
0 |
- |
|
1-8 |
- |
|
|
17 |
- |
m |
|
1-8 |
- |
|
|
18 |
- |
- |
C5 |
1-7 |
8 |
C |
|
19 |
- |
m |
|
1-8 |
- |
|
|
20 |
1 |
- |
|
1-8 |
- |
|
|
21 |
- |
m |
|
1-8 |
- |
|
|
22 |
- |
- |
C6 |
1-8 |
- |
|
|
23 |
- |
m |
|
1-8 |
- |
|
|
24 |
1 |
- |
|
1-7 |
8 |
D |
Figure 5 - The ESF Format
The chart shown in Figure 6 shows the differences between D1 through ESF formats. As most equipment today is either D4 or ESF, the data for D1 and D2 is displayed only for completeness.
|
Time Slots |
D1D |
D2 |
D3,D4,ESF |
|---|---|---|---|
|
1 |
1 |
12 |
1 |
|
2 |
13 |
13 |
2 |
|
3 |
2 |
1 |
3 |
|
4 |
14 |
17 |
4 |
|
5 |
3 |
5 |
5 |
|
6 |
15 |
21 |
6 |
|
7 |
4 |
9 |
7 |
|
8 |
16 |
15 |
8 |
|
9 |
5 |
3 |
9 |
|
10 |
17 |
19 |
10 |
|
11 |
6 |
7 |
11 |
|
12 |
18 |
23 |
12 |
|
13 |
7 |
11 |
13 |
|
14 |
19 |
14 |
14 |
|
15 |
8 |
2 |
15 |
|
16 |
20 |
18 |
16 |
|
17 |
9 |
6 |
17 |
|
18 |
21 |
22 |
18 |
|
19 |
10 |
10 |
19 |
|
20 |
22 |
16 |
20 |
|
21 |
11 |
4 |
21 |
|
22 |
23 |
20 |
22 |
|
23 |
12 |
8 |
23 |
|
24 |
24 |
24 |
24 |
Figure 6 - Channel & Time Slot Number Assignments
A Digital Cross-connect (DSX) consists of equipment frames (patch panels) where cabling between system components is connected. Each digital signal is defined for and handled by its own cross-connect. Thus, for example, DSX-1 is used to interconnect equipment operating with DS1 signals.
The pulse shape of a DS1 pulse is defined at the DSX-1 cross connect.
AT&T Publication 43801 describes the requirement of this pulse to drive from
0 to 655 feet of 22 gauge ABAM cable between the channel bank and the DSX-1. The
maximum time of reframe time is defined at 50 msec. Actually the DS-1 pulse is a
slightly relaxed version the DSX-1 pulse mask. Figure 7 shows the specification
(less template) of the DSX-1 signal and how it compares to the DS-1 signal
specification.
|
Functions |
DSX-1 |
DS-1 |
|---|---|---|
|
Line Rate |
1.54 Mhz +/- 200 Hz |
1.544 Mhz +/- 75 Hz |
|
Cable Length at DSX point |
ABAM/655 ft. |
6000 ft. |
|
Pulse Amplitude |
2.4 to 3.6 v. |
2.7 to 3.3 v. |
|
Receive Attenuation |
<10 dB |
15 to 22.5 dB |
|
Line Build Out |
Yes |
0.0, 7.5, 15 dB |
|
Max Successive Zeros |
15 (or B8ZS) |
15 (or B8ZS) |
Figure 7 - Comparison of DSX-1 Signals and DS-1 Signals
The ANSI standard T1.403-1989 is different yet again. Fundamentally the signals and the templates (signal shapes) are pretty much the same. Modern IC manufacturers have insured that their products meet all of the specs. When we are communicating to the CO or to the carrier we are using DS-1; when we are regenerating the signal after the demarc, we are using DSX-1.
It is important to note that the template of the DS-1 signal is bipolar. This means that a plus voltage, a zero voltage, and a minus voltage are important to the coding of the signal. The code which is used in T1 is call AMI for Alternate Mark Inversion. This means that if a "1" or Mark is coded as a positive voltage, the very next "1" must be a minus voltage or the result will be a Bipolar Violation (BPV).
Figure 8 shows a valid AMI sequence and a sequence with a BPV.
Figure 8 - Two AMI sequences
Notice that in the specification in Figure 7, there is reference to the
"Maximum Successive Zeros". One of the requirements of the coding
sequence and hence the signal shape of the DS-1 is that a "1" bit is
sent in order to maintain the timing synchronization. For example, a signal that
was sending all 0's would be a constant zero voltage line. Eventually the timing
of the system would be lost.
The requirement is that no more than 15 0's can be sent before a "1" must be transmitted. In telephone applications that was accomplished with bit 7. Remember, bit 8 is sometimes used for signaling so it couldn't be universally used. The human ear would never detect these slight variances in the lower order bits. In the case of sending data, using bit 7 and bit 8 for other than faithfully representing the data being presented for transport yields disastrous consequences. Thus a mechanism had to be developed for data only applications.
The easiest approach and a technique still in use in DDS is to make every bit 8 a 1 and to use only the lower 7 bits. This 7/8 mode yields 56Kbps instead of the standard DS0 rate of 64 Kbps. This technique also disallowed the use of signaling bits.
An improvement to this technique was developed known as B8ZS with stands for Binary Eight Zero Substitution. This technique takes advantage of BPV's in the data stream to be decoded as a signal.
With B8ZS coding, each block of 8 consecutive zeros is replaced with the B8ZS code word. If the pulse preceding the inserted code is transmitted as a positive pulse (+), the inserted code is 000+-0-+ (BPV's in position 4 and 7). If the pulse preceding the inserted code is transmitted as a negative pulse (-), the inserted code is 000-+0+- (again BPV's in position 4 and 7).
Figure 9 shows how B8ZS works.

This is the standard for "Clear Channel Capability". AT&T references it in Publication 62411 in Appendix B as CB144. It is part of the ANSI T1.403-1989 standard as well.
Now for some discussion on ABAM cable. This is the cable that is called out
in the DSX-1 spec and is a physical cable that was manufactured by AT&T.
Generally it is a cable that has unshielded twisted pairs with a wire size of 22
AWG. Some authorities suggest that it is pulp insulated while others suggest
that it is plastic insulted. In any event, ABAM cabling, per se, is no longer
available. Modern cable manufacturers, however, especially those active in
EIA-568, have developed cables with specific categories or levels.
Category/Level 2 cable is adequate for the T1 data rate and has the following
characteristics:
Several manufacturers make this cable type. A summary of the Category/Level types per RS-568 is listed in Figure 10.
|
LEVEL |
SERVICE TYPE |
SPEED |
|---|---|---|
|
1 |
POTS (plain old telephone service) |
n/a |
|
|
RS-232/RS-562 |
19.2 to 115.2 Kbps |
|
|
T1, Fractional T1 |
64 Kbps increments |
|
|
ISDN Basic Rate |
144 Kbps |
|
|
RS-422 |
up to 1.0 Mbps |
|
2 |
IEEE 802.3 1BaseT |
1.0 Mbps |
|
|
IBM System 3x/AS400 |
1.0 Mbps |
|
|
T1 |
1.544 Mbps |
|
|
ISDN Primary Rate |
1.54 Mbps |
|
|
IBM 370 |
2.36 Mbps |
|
|
IEEE 802.5 |
4.0 Mbps |
|
3 |
Wang |
4.3 Mbps |
|
|
IEEE 802.5 10BaseT |
10.0 Mbps |
|
|
IEEE 802.5 Token Ring |
16.0 Mbps |
|
4 |
IEEE 802.5 Token Ring |
16.0 Mbps |
|
|
New Arcnet |
20.0 Mbps |
|
5 |
X3T9.5 TPDDI |
100.0 Mbps |
Figure 10 - New Cable Types (Proposed EIA-568)
The discussion of connectors sometimes becomes confusing as there is a difference between "de facto" standards, things used in products, and specification. AT&T specify that the Network Interface (NI) should be a subminiature 15-pin female connector with the following pin-out:
| 1 | Send Data (tip) |
| 2 | Reserved for network |
| 3 | Receive Data (tip) |
| 4 | Reserved for network |
| 5 | Not defined |
| 6 | Not defined |
| 7 | Not defined |
| 8 | Not Defined |
| 9 | Send Data (ring) |
| 10 | No connect |
| 11 | Receive Data (ring) |
| 12 | No connect |
| 13 | No connect |
| 14 | No connect |
| 15 | No connect |
AT&T Publication 62411 further states that "in such cases where ISDN
standards need to be met, an 8 pin mini-modular connector is recommended"
with the following pin-out:
| 1 | Transmit (ring) |
| 2 | Not Used |
| 3 | Not Used |
| 4 | Receive (ring) |
| 5 | Receive (tip) |
| 6 | Not Used |
| 7 | Not Used |
| 8 | Transmit (tip) |
To complicate the matter, ANSI T1-403-1989 specification calls out for "one
of four Universal Service Ordering Code (USOC) connectors (RJ48C, RJ48X, RJ48M,
and RJ48H)" with pin assignments as follows:
| 1 | Receive (ring) |
| 2 | Receive (tip) |
| 3 | Not Used |
| 4 | Transmit (ring) |
| 5 | Transmit (tip) |
| 6 | Not Used |
| 7 | Not Used |
| 8 | Not Used |
As it goes, the above pin-out and connectors is also the "de facto" standard vis-à-vis how currently available hardware is configured.
Well, then, what do we do with these DS-1/DSX-1/T-1 signals? There are several applications and specific equipment that can be applied.
The most important issue to see is that there can be T1 networks that are customer owned and T1 networks that use the AT&T Accunet T1.5 system. The applications will be the same but the constraints on the equipment are more stringent using the AT&T connection.
There are three levels of DACS compatibility. The first level is DS-1 and is at the full T1 rate. The second level is "bundled" or 1/4 T1 level. This allows the customer to utilize Customer Controlled Reconfiguration or "fanout" at the CO (central office). The third level is at the 64 Kbps or DS-0 level. What happens is a single T1 signal is generated using channels a and b and goes to the CO. The CO splits this into two T1 trunks one carrying channel a and the other carrying channel b. The device the performs this function is called a DACS. DACS may also be configured with a topology such as a ring topology. If one of the trunks goes down, the data will be reconfigured to go over the standby trunk. Almost all DACS are owned by the telcos
As we mentioned the T1 signal must somehow be split into the 24 separate and distinct voice channels. When this is done, it is still in the digital form. The codecs must then convert the digital signal (per channel) into analog signals to be sent on the subscriber loops. Again, most Channel Banks tend to be owned and operated at the CO's (Central Offices). Since deregulation in the 1980's, more T1's are owned by users, as telephone carriers continue to reduce the cost of the local loop (the wires from the central office to the customer premise).
Clearly the intended use of T1 was to bring in as many telephone lines using voice as possible through a digitized technique (PCM Pulse Code Modulation). Tie lines between PBXs account for many private T-1 network applications. This is supported through 2 and 4 wire E & M (Ear and Mouth) signaling techniques through the T1 Mux. A 2w FXS (Foreign Exchange Subscriber) function (dedicated line to a distant CO) and 2w FXO (Foreign Exchange Office) function (the CO version) can also be supported by the T1 trunk. In the latter mode, the T1 line acts as an "extension cord". The primary way in which customers use this function is through the T1 Multiplexor.
This may be the easiest to explain. A DS-1 comes from the phone company to the customer. This line must be given the proper termination, line protection (vis-à-vis FCC Part 68), and message handling capability. In the old days, the phone company supplied this equipment but today this probably will be CPE (Customer Premise Equipment). The output of the CSU is the DSX-1 signal. The most common CSU is found in a T1 Mux however they can stand alone with various added functionality.
The bipolar output of the CSU can be connected to a DSU (Digital Service Unit) which converts the bipolar signals to unipolar and vice versa at the data rate gleaned from the bipolar signals.
The DCB T-Driver, for example, as it stands today, is a DSU. It takes unipolar data from the terminal and coverts it to a DS-1 signal. In many ways it also acts as a CSU and its transition to a CSU/DSU is quite possible. AT&T Pub 62411 requires that a CSU perform the following functions:
The regeneration part is part of the present T-Driver functionality. Loopback is commanded from the Carrier in one of two ways:
As the FDL is already being used in T-Driver, it would be rather straightforward to incorporate the appropriate responses to the command structure of the loopback from the carrier. The interface is already surge protected and meets FCC Part 68. The conclusion is that we have with relatively small impact an "ESF CSU" in the T-Driver product that can connect directly to the carrier. To incorporate an "SF CSU" which is still quite prevalent in use with D4 channel banks, would be a more significant undertaking requiring hardware and software changes.
As a matter of note, DDS (Digital Data Service) also requires a CSU but most units are sold as a CSU/DSU with a V.35 connector right on the device. T-Driver will be unique in that regard in that it is the same concept as the DDS CSU/DSU only running at much higher speeds. In this situation T1 will be solely used for data.
This is actually a family of devices dedicated for customer use. They are normally T1 or fractional T1 TDMs which comply with format constraints , DACS interfaces, and often have an optional CSU. Their purpose, depending on the number of ports, is to allow transmission of data, image, and voice form many different sources of a single network link.
Many T1 Muxes are also Subrate Data Muxes (SRDMs). By this identification they are able to accommodate synchronous data rates of 2.4, 4.8, 9.6, and 19.2 Kbps. Asynchronous data rates are also allowed in some devices. SDRM operates per DS0.
Since T1 muxes are also DACS compatible at the DS0 level, Fractional T-1 service is also compatible with the devices. They also comply with the D4 channel bank requirements of bit density, zero density, and the provision of clear channel. FT1 is like SRDM only at the DS1 level. Hence, data may be at multiples of 64Kbps.
Also many T1 Muxes allow for the integration of the AT&T Switched 56 service. These are important month-end transfers, CAD/CAM files and teleconferencing.
The FT DSU/CSU's have a DS-1 output signal, and are FCC registered DSU's. They take data at a configured speed via an RS-530/V.35 interface and convert the data to a T-1 data stream. The format of the data is can be D-4 or ESF. The transmitter is configured with a selectable signal attenuator (LBO) of 0, 7dB, and 15 dB per AT&T spec. The FT series is available in a single channel units (FT-1), two channel unit (FT-2) and a 4 channel unit (FT-4). Each port can be configured to use from 1 to 24 of the DS-0's (56 or 64 Kbps each DS-0). The FT-2 and FT-4 units also have drop and insert capability.
T-Extender is a T1 repeater designed to AT&T specifications. This device takes a DS-1 signal and regenerates it as a DS-1 signal. T-Extender can have the DSX-1 output of T-Lan as an input signal and T-Lan will also accept and decode the output of the T-Extender. T-Extender, being a signal repeater, is not constrained by any formatting. For example, a BPV is passed through just a readily as a normal signal. The output of T-Extender is -4 dBdsx and is fixed. This is -4db from the allowable power as defined in the Repeater Specification, AT&T Publication TA24/CB113 and was done to simplify the circuit. The product has a robust receiver and therefore should have no difficulty in going repeater to repeater nearly 6000 feet on 22AWG solid, shielded twisted pairs.
T-Driver, as we discussed, is a DS-1 output signal, and acts as a DSU. It takes data at a configured speed via a V.35 interface and converts the data to a T-1 data stream. The format of the data is ESF and the FDL is used for command the remote units using a proprietary protocol. One's density is maintained by using B8ZS coding. The receiver is enhanced with an ALBO (Automatic Line Build Out) which is also used in T-Extender. The transmitter is configured with a selectable signal attenuator (LBO) of 0, 7dB, and 15 dB per AT&T spec.
A simplified equation for the definition of dBdsx is the following:
dBdsx = 20 X log (.167 Vp-p measured)
where "Vp-p measured" is the peak-to-peak measurement of the voltage between tip and ring. For example...
If there is a 0.5 volt positive voltage on tip and a 0.5 volt negative voltage on ring...
The peak-to-peak voltage measurement is 1.0 volts. Using the equation,
dBdxs = 20 * log (.167 X 1.0)
= 20 * (-.777)
= -15.5
Notice that tip and ring signals are inverted. When a 1 is sent one line (tip, for example) will be a positive voltage and the other (ring, for example) will be a negative voltage. When 0's are begin sent, both lines are at 0 volts. Since T1 is AMI or alternating, the next 1 will have the voltages reversed.
Many specifications give the "pulse amplitude" rather the dBdsx.
This parameter is the positive voltage, measured from zero, of a 1 being sent.
In other words, it is half of the peak-to-peak voltage. As a note of interest,
the T1 pulse is not specified as necessarily symmetric. AT&T Pub 62411 states
that the maximum + voltage is defined as 3.0 +/- 0.3 volts while the maximum -
voltage is its absolute value (without sign) and must be within 0.20 volts of
the + voltage but no less than 2.7 volts or greater than 3.3 volts.