Legacy Telephony FDM to TDM (PCM/DS1-framing-coding and alarms)




Though telephony is now in the VOIP world today, understanding were it evolved from  can make for a better modern Telephony-collaboration Engineer
FREQUENCY DIVISION MULTIPLEXING or FDM is a good place to start to understand more in depth what it is that is being converted to packets. FDM is simply taking a 40 up to 4000 HZ voice channel and modulating it to a higher channel frequency assignment. Much as Cable TV companies have stacked TV channels, so did the phones companies.  They started with the base-band frequency of 40-4000 HZ. Then Channel Two would have its base-band range modulated to start at 4000 HZ to 8000 HZ, Channel Three would be from 8000 HZ to 12000 HZ. This would continue until it reached 96000 HZ, which would give a 24 two-way voice channel over two-pair wire one pair for receiving and one pair for transmitting (thus the term 4 wire circuit). This is purely an analog transmission and is subject to problems that Analog circuits have and that is the noise and other distortions that are induced every time the signal needs to be re-amplified. I had to work on some Analog Radio before it was all taken out of service in the mid 90’s. Not much of it exists anymore but there s still some of it running somewhere in the world.
In the late 1950’s and early 60’s Bell engineers found a way of converting signals to a digital format from analog signals for transmission, with the digital conversion signal-to-noise ratio significantly improved over that of the analog FDM systems. FDM was like the old record players and analog tape of that time that reproduced sound from magnetic tracks that create audible sine waves. Waves from these record and tape players needed a boost and went thru pre-amplifiers and amplifiers that add some noise along the way. This early T-1 technology is basic building block for the plethora of digital audio technologies we have today . With the implementation of digital conversion the transmission is achieved over longer lengths of existing copper wire-pair, often this type of transmission was more economical than the FDM method. For FDM to be used on long distance hops it needs to be transported on coaxial cables and/or Microwave Radio. Digital signals can be transported over twisted pair at greater distances than Analog FDM. This originally had a great cost benefit. But even greater digital bandwidth can be achieved if it can be put over coaxial cable, or microwave The signal to noise ratio is greatly improved with either media. Time Division Multiplexer (TDM), also called a digital channel bank was developed from this new technology.







For the 4,000 Hz voice channel in the FDM system to be converted to a digital signal and accurately reflect information in that 4000 Hz spectrum a sampling rate of twice the frequency or 8000 Hz is required. This is dictated by the Nyquist sampling rule that digital sampling of an analog wave is twice the frequency. Figure 1 is a representation of a analog signal that is sampled and then Time Division Multiplexed into Pulses that are synchronized to a clock pulse, each sample is quantified and coded into 8 bit binary PCM words. Since the dynamic range of a voice channel is very broad ( 60 Db which is equal to 1000 to 1 ratio of amplitudes possible) this range is divided into 256 quantum steps which reduces that range. Since the human ear cannot detect many of these minute amplitude changes it would be a waste of bandwidth to pass them on. So the next step is to compand or place emphasis on voice channel intensity were the amplitude levels will be perceptible. Companding changes the normal relationship of speech amplitudes by applying more gain to weak signals and less to strong signals. By doing this the voice signal can be compressed and still offer good quality reproduction range.




Finally allocation is made for equal time-slots to each of the 24 voice channels synchronized to a clock signal. Twenty-four channels are serialized into 24 time slots one after another using two wire pairs, one for each direction. Sampled at 8,000 samples a second, this produces a data rate of 64,000 bits per second for a single voice channel. This is known as Digital Signal level Zero (DS-0) commonly called a channel on a T-1 or just a ds0.




The process Pulse Code Modulation is completed with the final T-1 signal of twenty-four 64,000 bit per second channels, each analog channel has now been sampled and coded into a digital wave train, by the Line Interface Unit in the channel bank or some other type of Digital Signal Unit if channel bank is not being used. Each channel is given 8 bits per time slot, with one sample for each channel resulting in 8 x 24 = 192 bits. Now if these bits were to be sent down the transmission media, they would be unrecognized at the other end without some additional information saying what all these on and off signals are. Framing is used to differentiate all this data with one additional bit added as a framing or synchronization bit. This bit separates the data into organized partitions of information; this additional frame increases the size to 193 bits. A T-1 transmits at a bit rate of 1.544, this obtained by multiplying 8,000 samples per second x 192 bits per sample. This yields an overall bit rate of 1.536 Megabits/second. With an additional 8,000 framing bit overhead= 1.544 Megabits ( or 193 x 8,000 = 1.544) combine with the digitized voice. This bit rate is called Digital Signal level one (DS-1). The DS-1 signaling rate is the fundamental building block upon which the North American digital hierarchy was formed. In 1961, after beta testing the T-1 trunks were shortly set up between exchanges in Chicago. The original use of the T-Carrier System–an AT&T service product name–was to carry twenty four-voice channels between exchange offices. These systems (Figure 4) had terminal equipment at each end with many regenerative repeaters. A repeater does not amplify the signal, but it literally reproduces the signal again in its original form. These repeaters are spaced at 6,000’ intervals between the terminating points. The transmission lines used 16 to 26 gauge wire pairs. The lines and the repeaters are called a span.



Load coils were put in every 6,000′ on the twisted pairs. Since access was available at these intervals, this became the obvious place for repeating generators. The load coils in the Service Area Interface enclosures where removed and repeaters were installed. Experiments in the late 1950s and early 1960s found that 1.5 Megabits per second was the maximum reliable bit rate that could be obtained on standard twisted pair with regenerative repeaters every 6,000 feet.



Each end is terminated with a Network Interface Unit (NIU) which commonly called a smart jack that provides a Line Build Out (LBO) that compensates for the final cable distance between the last repeater and span termination. The Smart Jack also has relay circuit that can respond to loop code from remote test equipment and provide a loop back to the test set. The Smart Jack also provides a demarcation point between the Telephone network and the Customer Premise Equipment (CPE) A Channel Service Unit (CSU) is usually installed after the smart jack. The CSU also has loopback capability and is typically programmed to loop up on a different code than the smart jack. The CSU also has Line Build Out circuitry that can be adjusted to provide that right amount of line equalization. This is very useful when the CPE equipment is several hundred feet from smart jack. Digital or Data Channel Units (DSU=s) are the final termination of the transmission circuit.



A channel bank can be the DSU were it is the digital to analog conversion point. Or it can be a digital trunk card in a PBX were PCM is used for switching many lines. The DSU can also be in the same unit with a CSU and be the DTE equipment for a Wide Area Network. Basically the DSU can be a Digital modem that can modulate data that is in another format into a PCM signal. One thing the DSU is not is smart jack and a demarcation point of the Local exchange. I know many PBX technicians that think the smart jack is a DSU. It is probable for more than one carrier to be involved in a T-1 span if is used in an intrastate or LATA connection. When LEC hands off the span to Long distance Interstate eXchange Carrier IXC they will have a point of presence (POP) in the Local Exchange Carrier LEC Central Office and will issue a Carrier Facility Assignment for the LEC to cross connect to the LEX It is at this point that I would like to point out where much of the confusion is with the term T-1. The term “T-1,@ originated by AT&T, called for a very specific type of physical equipment and bit rate: digital re-generators at 6,000 foot intervals, and a bit rate of 1.544 Mbits/sec and most particularly cable pairs designed to give better capacitive properties in the dielectric. These cable pairs were the beginning roots for the development of category 3 and now category 5 twisted pair that is commonly called cat 5 cable, which has up to 100 MHZ bandwidth. Today, when someone says T-1 they most likely are referring only to the DS-1 rate rather than to the cable system.


The hierarchy of the Digital signal service levels and how voice channels go up with the bandwidth capacity. This chart reflects how digital service designation progressed as the mediums the Phone companies used created more bandwidth. To note in today’s world this chart is obsolete as we are using CAT-5 and CAT -6 cables along with the EIA=TIA 568 A-B pin-out terminations.


Digital service number Number of Voice




Bit Rate


Typical Transmission


DS0 1 NONE 64K Unloaded twisted pair
DS1 24 D channel bank

24 analog inputs

1.544 T1 paired cable
DS1C 48 M1C (2 DS1 inputs) 3.152 T1C paired cable
DS2 96 M12 (4 DS1 inputs) 6.312 T2 paired cable
DS3 672 M13 (28 DS1inputs) 44.736 Microwave, 75 ohm Coaxial cable, Fiber Optics
DS4 4032 M34

(6 DS3 inputs)

274..176 Microwave, Coaxial cable, Fiber Optics



In many installations there are typically combinations of 64K (DS0) Voice and Data channel modules mixed in a channel bank like the Telco channel bank in Figure above with all of the output/input=s of each card being attached to a PCM bus that multiplexed together by Line Interface Unit or LIU. It is the LIU that is attached directly to the T-1 line.



There are a variety of cards that can be installed in the channel bank. One type channel card called either a FXO (Foreign eXchange Office) or FXS (Foreign eXchange Station) card has hybrid circuits that convert the 2 wire circuit into 4 wire circuits that create separate transmit and receive paths before the signal is introduced to the PCM Bus. The use of the FXO would be at a Central office or a PBX site. And the FXS is used at a remote side where card output is cross connected to a pair to the home, business or other station.



With the framing of a 64 kbit/second DS-0 signal can be multiplexed onto a T-1 carrier with other voice or data channels, but is not necessary that a T-1 circuit be channelized specifically in 64 kbit slots. The channelization of voice or data signals is an application of a T-1 carrier and the use of a T-1 carrier does not require the channelization of data. The T-1 can also facilitate asynchronous types transmissions like Frame relay. This type of data transmission was developed with ARPANET which is predecessor to the Internet. With TDM with each tick of the clock data/voice bits are sequenced into a time slot for each channel. This becomes a waste of bandwidth when the different channel slices have no bits to present for the time slot, and the time slot goes empty. Then came Frame Relay and Asynchronous Transmission Mode or ATM does is shift bits into these unused timeslots. Essentially they swap bits around and borrow space filling up every time slot with reassembly at the other end.  by late 80’s and early 90’s Online services like CompuServe came into service using the ITU X.25 protocol with local servers and modem banks connected to 64 kbps channels that performed data packet switching between connected modems to the host.  By the early 90’s another use of DS1 bandwidth is the ISDN Primary Rate Interface (PRI). Inside North America and Japan, this consists of 24 channels, usually divided into 23 B channels and 1 D channel, and runs over the same physical interface as T1. Outside of these areas the PRI has 31 user channels, usually divided into 30 B channels and 1 D channel and is based on the E1 interface. It is typically used for connections such as one between a PBX (private branch exchange, a telephone exchange operated by the customer of a telephone company) and a CO (central office, of the telephone company) or IXC (inter exchange carrier, a long distance telephone company).
These asynchronous modes use the entire bandwidth of 1.536 Mbits/second T-1 or mutliple T-1 carrier circuits.
T-CARRIER FRACTIONS.  By the 90;s Companies started to lease T-1 lines and were terminated at a customer’s facility via a Digital Service Unit (DSU) and when required a Channel Service Unit (CSU) on the premises. The Digital Service Unit is usually the Local Exchange’s equipment that they provide at the demarcation point A customer has access to the payload data stream of 1.536 Mbits/second. Within the last few years, the introduction of fractional T-1 (FT1) circuits have provided direct access to portions (fractions) of a T-1 circuit on a channelized basis. This has allowed a larger customer base access to T-1 facilities.
The AT&T and ITU-U Standards
Originally, AT&T defined and maintained the standards used within the public telephone network. Since the divestiture of AT&T in 1984, several organizations now maintain similarly, but as usual not exact standards. These three standards are: AT&T’s TR. 54016 and TR. 62411; ANSI’s T1.403 and T1.107; and Bell Communication Research’s (Bellcore) TR-TSY-000194, they provide service descriptions, performance and maintenance parameters, format specifications,

and electrical characteristics of T-1 equipment and signals. Nevertheless, all this will probably change As Wide Area communications starts migrating to Asynchronous communications and all these specifications became ancient history in the next decade. Asynchronous Transmission Mode or ATM will be taken be discussed in the next chapter

T-1 transmission protocols and Troubleshooting

Alternate Mark Inversion or AMI

The first digital transmission was Morse code over the telegraph line in which the telegrapher would key in marks and spaces. Which is were the “mark” in Alternate Mark Inversion comes from. AMI is a line coding method by which the logical bits of data are represented over a network line. Alternate Mark Inversion (AMI) is a bipolar return-to-zero code and is the basic line coding procedure used for a T-1 carrier. In Chapter One a digital signal is defined as “1″ equaling the common 5 volt square pulses (though it can be from 3.3 to 12 volts) and the “0″ with no voltage. The bipolar signaling technique still transmits the “0” as zero volts, but the “1” can be either a positive or negative pulse. All 1’s will alternate polarity to the previous “a 1″ pulse despite the number of intervening spaces. A sample AMI signal is shown in Figure 1. The reason for doing this is to create more of an analog type square wave, the alternating pulses eliminate any stray DC pulse components over the transmission medium. The DC component would ultimately lead to data errors at the receiving end. The alternating cycles of 1’s force each pulse to return to zero before the next period. This reduces what is known as inter-symbol interference and aids the timing circuitry in the regenerative repeaters or amplifiers in accurately preserving the data information in the T-1 span.


T-1 lines uses a clock that provides a synchronous transmission method by which all data can be processed. Synchronous methods require the establishment of a time base (clock) at the receiving end to identify the proper times to sample the received signal. As can be seen from figure 1 the timing information is embedded within the actual signal with AMI coding. That is, the clock positioning is determined by alternating pulses of the 1’s. Just as the original signals are sampled and multiplexed into the data stream, at the other end the data is picked out of the stream by sampling also. This sampling period is inserted between transitions of the data and empty intervals created when a data transition does not occur. Because AMI uses alternating polarities for encoding 1s, strings of 1s have strong timing components. Strings of 0s, however, contain no timing information and must therefore be limited by the source. A low density of 1s within a data stream can lead to the addition or deletion of a bit commonly called clock slips. Also timing jitter (the cyclic offset of bits from their expected positions in time) is also a possible problem. This can contribute to a higher bit error rate and requires that a certain minimum amount of these bipolar signal transitions happen. The repeaters and line terminating devices will maintain timing as long as no more than 15 consecutive 0s occur in a row,this follows specifications for T-1 line standards. When the T-1 circuit is transferring channelized voice, rarely will two adjacent channels produce 16 consecutive 0s. On those few occasions when this does occur, the Least Significant Bit (LSB) of the second channel in the 16 bit steam is simply overwritten with a 1.This one bit error is imperceptible to the human ear. For transferring data, however, the probability of consecutive 0s is much greater and a single bit error is unacceptable. Furthermore, since the least significant bit in every sixth frame is used for in-band signaling (to be discussed later), these bits are unusable for transferring data. One bit in every eight (the LSB) is set to 1 to satisfy the ones density requirement. This is why the DS-0 channel is limited to 56 Kbits/second for data transmission (as with the DDS service) and T-1 data rates are 1.344

kbits/second (24 channels x 56 kbits/second), seven-eights (87.5 percent) of the 1.536-Mbits/second rate.
This method was developed by AT&T Bell Laboratories Research in the mid 1980’s and was named B8ZS, for Bipolar 8-Zero Substitution. Since the seven of bits are actually used for data in a channel, B8ZS takes advantage of the eighth channel bit during data operation. This alternate method was developed to improve the problem created with long strings of 0s. Unlike the original AMI this coding that limits zero transmissions, this scheme allows a terminal to send all zero transmissions along with the 1’s.In January of 1989 this service, was first offered by AT&T, and called Clear Channel Capability (CCC).
with B8ZS coding, each string of eight consecutive 0s is removed and is replaced with an eight-bit B8ZS code containing bipolar violations in the 4th and 7th bit positions of the substituted code. The equipment on both sides has to be set to the same coding scheme or the BZ8sS will be detected as bipolar violation errors creating bit errors, or worse yet alarms and a shut down of important 800 voice channels to the PBX that You are maintaining. Just as modems have to have the handshaking right to communicate, so it is with T1 transmissions.

D4 and ESF Framing
As mentioned earlier the frame supplies organization to data that would otherwise be meaningless.,
There are two formats in use today with T-1 circuit, one of two framing formats is D4 and the other is ESF or Eextended SuperFrame.

D4 Framing (Super Frame)
A Superframe (D4) format consists of 12 frames of 193 bits each for a total of 2,316 bits. Each 193 bit frame consists of 192 voice/data bits followed by one framing bit (the F bit). At 8,000 samples per second, the 192 data bits produce a payload data rate of (8,000)(192) = 1.536 Mbits per second. Adding the 8,000 framing bits per second results in the DS-1 bit rate of 1.544 Mbits per second.

The 192 voice data bits are 24 consecutive 8-bit voice channels when the service application is channelized voice. They can also be defined by the user when the service application is for user transport. The F bit takes on the consecutive values of “100011011100” within a superframe. Since it is highly unlikely that any random data will sustain this pattern for any length of time, the pattern identifies the start of a superframe. The superframe’s structure and format are detailed in Figure 4

As seen in figure 4, the superframe actually has two other frame patterns within, the first is called the Terminal Framing pattern (Ft) consisting of framing bits in the odd numbered frames. This pattern alternates between 1s and 0s. Terminal framing is for frame synhronization it insures that the data streams are routed to the correct channel termination within the channel bank. The second pattern, called the Signal Framing (Fs) or Multi-Framing (Fm) pattern, identifies those frames in which signaling bits, are transmitted for a voice channel. The signal framing is primarily used in voice channels and typically is not used with data transmission. Signal framing creates a “A” and a “B” signalling channels within the superframe The sequence can consists of framing bits in even numbered frames is, for example, “001110”. By examining Figure 4, it can be seen that a 0 to 1 transition of Fs in the even-numbered frame signifies an A signaling frame. Correspondingly, a 1 to 0 transition in Fs signifies a B signaling frame. Frame acquisition begins by first identifying the alternating bit sequence and then by identifying the signaling frames. Rather than carry signaling with every time slot, the channel banks put signaling information in every sixth frame. This requires synchronization with the Signal Framing sequence so that the sixth and twelfth frames of the superframe structure can be identified. In these frames, the least significant bit of voice data in each time slot is over-written with a signaling bit. This process is called bit robbing. The receiving channel bank interprets these positions separately. With two signaling bits in each superframe (‘A’ in the sixth frame, and ‘B’ in the twelfth frame), up to four signaling conditions can be transferred per frame. In most situations, both A and B are made the same: 1 for “on-hook” and 0 for “off-hook”. These two supervisory states and the sequence of transitions between them can control or indicate disconnect, busy, dial pulses, etc. For example, the off-hook indication would be signaled by setting A and B to 0. Dialing could be indicated by transitioning A and B between 0 and 1 within a specified period of time and on-hook would be indicated by setting A and B to constant 1s.
Channelization and signaling are only pertinent when terminal devices exist on each



end of a link which recognizes this structure, such as D4 channel banks. When data is “carried” over a T-1 Carrier, the channelization and signaling bits are not neededbut could utilize some other procotocol or scheme
Extended Super Frame (ESF).
The ESF was delveloped to provide a means of end-to-end non-intrusive performance monitoring and maintenance functiont. The ESF is defined by the following standards: 1.)The ANSI T1E1 committee with T1.403. At the same time, Bell Communication Research (Bellcore), which is sponsored by the seven regional Bell holding companies with there own specifications consistent with ANSI T1.403. 2.) An older standard called TR54016 implenmented during the 1980’s, a lot of older equipment was installed with this standard and it may be that both standards will be around for a few more years.
ESF Format
The ESF format doubles the length of a superframe to 24 DS-1 frames. As previously defined, a DS-1 frame consists of 192 data bits and one framing bit, referred to as the F-bit. The 24 F-bits in each ESF are partitioned into the following: 1.) Six of the bits are used for synchronization called the Frame Pattern Sequence as shown in Figure 5. The FPS sequence, that provides the mechanism to identify an extended superframe, is “001011”. 2.) Six are used as a Cyclic Redundancy Check (CRC6). The CRC is generated by the CSU or terminal equipment generating the data, a data error occurring anywhere in the path will be detected by the CRC check at the receiving terminal equipment. CRC is a popular standard for many transmission protocols in the computer world like Zmodem and Xmodem used for file downloading and file compression programs like Pkzip. 3.) The remaining 12 bits are used to create the Data Link (DL) a 4 kbit/sec outband data channel. The The Channel Service Unit and the Digital Service Unit facilitates use of the ESF Data Link (DL) for performance monitoring and maintenance commands along with customer defined messages. Thru CSU/DSU the DL is utilized for maintenance messages and acts upon these messages. The messages include commands to initiate loopback functions and requests for a variety of error and performance statistics accumulated by the either the DSU or CSU. Basically the DSU is used by the carrier that is providing the leased line ( I.E. US West, Bell Atlantic) span for there diagnostic purposes. The CSU would normally be for the specialty service, like a long distance carrier ( AT&T, Sprint and MCI ) providing long distance service to a company PBX. Or it could be a tie trunk that would supply voice channels between two PBX’s on 12 channels and Data on the other 12 channels. In this case the CSU’s would be used by the company with the 2 Pbx’s There are Two standards when it comes to how the CSU is used, one is the TR54016, were the CSUs store up to 24 hours of performance statistics and communicate that data to the network Line Monitoring Unit via the Data Link. The Line Monitoring unit or Line Analyzer is connected to an isolated test point that can either be accesed through the CSU or the DSU. The other standard is T1.403 which requires the CSUs to transmit performance data onto the network with a DL message called a Performance Report Message (PRM). The PRM is transmitted each second and contains performance data from the previous 4 seconds. The TR54016 data link consists of a link layer protocol conforming to the popular SX.25 protocol that is primarily used on 64kbps Wide Area Network. Although both the AT&T and ANSI standards specify these equivalent protocols, the implementation between the two differs, they are not directly compatible.


Performance Monitoring and Error Analysis
The specific reason for the development ESF was to provide a mechanism for better trouble shooting through performance monitoring and error analysis through the DSU/CSU. Many alarm states in table 2 can be found as alarm lights on the DSU/CSU.
Table 2

Loss of Signal (LOS) Condition. A LOS is declared when the NCTE has determined that 175 ± 75 successive pulse positions with no pulses of either positive or negative polarity have occurred. This light will be found on both DSU and CSU and it will generally indicate a break in either the send or receive pairs or whatever media that may alos be used in the T-1 span

Loss of Frame (LOF) / Out of Frame (OOF) Condition. A LOF condition is declared when either the DSU/CSU senses error in the framing pattern. A common problem that can cause this alarm to happen can interfering electromagnetic waves that can ingress into the cabling or the CSU or dirty power causing spikes interupting the regular framing sequence. It is important that the cabling have proper shileding with a ground drain on one end, this is important when the room can have a lot of RF generating equipment. For example a PBX is the equivalent of a computer with a Super time switching circuitry that can generate high frequency audio and low frequency RF typically in the AM area. These spurious waves can ingress into the into the cabling and possibly overwrite framing bits. Most shielded cabling usually will have one common shield for the two send and receive twisted pairs, which will not always be enough. I like the idea of using a separate shield for each send pair and receive pair, this can provide much better rejection outside interference. This could also be faulty wire connections

CRC6 Error. This ESF error occurs when the CRC6 field calculated by the CSU or the DSU, does not agree with the CRC6 field contained in the CRC6 field in the DS-1 signal. This problem could be caused by everything that was discussed with Loss of Frame, but most likely could a loose on damaged connection or splice within the span

AIS CFA / Blue Alarm State. A receiving terminal enters this state upon detection of an AIS signal.

Red CFA. A “Red” CFA is an alarm state existing at a receiving terminal, resulting from that terminal’s detection, on an incoming line, of a system failure (IE. LOS).

Yellow CFA. A “Yellow” CFA alarm state is activated at a terminal as a result of that terminal’s detection of a “Yellow” alarm signal received from a terminal which is in a “Red” CFA alarm state.

Alarm States and Signals
The AT&T specification TR62411 and the ANSI specification T1.403 defines several “events” which generate alarm signals over T-1 circuits. These signals initiate alarm states within the terminal equipment. The alarms, as a class, are referred to as Carrier Failure Alarms (CFAs) and along with several associated error events are defined in Table 3-I. The CFAs serve as indicators to problems detected on a carrier signal. Upon the detection of certain CFAs, terminal equipment must transmit alarm signals over the network to notify downstream equipment of the error condition.
One of these special signals is called an AIS (Alarm Indication Signal), also known as a Keep-Alive or Blue Alarm signal. The AIS is transmitted in lieu of a normal signal upon the loss of the received signal or when any action is taken that would cause a signal disruption. Figure 3-1 depicts where an AIS signal would be generated. The signal is used to maintain transmission continuity and to notify downstream receiving terminals of a transmission fault located either at the transmission equipment or upstream from the transmission equipment. The AIS is an unframed, continuous alll 1s signal.



The second type of alarm signal, shown in Figure 3-1, is called a Yellow Alarm signal or Remote Alarm Indication (RAI) signal. When a DS-1 terminal determines that it has lost the incoming signal or identifies any type of system failure on an incoming line, the terminal enters the Red Alarm CFA state and transmits the RAI signal in the outgoing direction, that is, the direction towards the origin of the lost signal. A Yellow alarm state is activated at a terminal as a result of that terminal’s detection of the RAI received from a terminal which is in a Red CFA alarm state. When operating using the superframe format, the Yellow CFA Alarm signal is generated by forcing the second bit to 0 in all channels of the DS-1. For the Extended Superframe format, a repetitive 16-bit pattern consisting of eight 1s followed by eight 0s (1111111100000000) is continuously transmitted for a minimum of one second, over the ESF data link.
Performance Parameters
The alarm conditions described above provide adequate indication of the state of an operating circuit but provide no information as to the quality of an operating circuit. For proper maintenance and management of expensive and critical T-1 circuits, performance monitoring is a necessity. The quality of T-1 circuits is dependent upon the basic electrical characteristics of the T-1 signals. One measure of these characters is Bipolar Violation.

Bipolar Violations
The use of an alternating bi-polar coding format as a line code results in a number of unused or illegal signal states. For example, since the pulses used to represent 1s must be opposite in polarity, that state corresponding to the same polarity as the previous pulse is not allowed. The inefficient use of this code space — that of a ternary code in which at any one time only two of three possible values are valid — provides the capability to monitor the quality of a line with no knowledge of the information being transmitted. Since alternate 1s must have opposite polarities, the detection of two successive pulses of the same polarity (other than the intentional violations inserted by the B8ZS coding algorithm) implies an error. This event is known as a bipolar violation (BPV). The occurrence of a BPV necessarily implies an odd number of bit errors. Bipolar violations are used as a type of parity check. T-1 terminals monitor BPVs and indicate an alarm condition if the number of BPVs exceeds a specified threshold.
As a performance indicator, however, the BPV is limited. Because channel multiplexors, digital cross-connects (DSXs), or other T-1 ports are sources of bipolar signals, they must not generate bipolar violations. Other types of equipment, such as T-1 ports on fiber optic multiplexors, must correct any violations received because the fiber does not carry bipolar signals. A typical T-1 circuit may contain several bipolar signal sources. A CSU or other CI can monitor bipolar violations as a means of quality monitoring. The disadvantage to this type of monitoring is that BPVs will reveal only the presence of errors in the last segment of a circuit, from the last bipolar pulse source to the monitoring unit. Nevertheless, the BPV is the only non-obstructive method for observing circuit performance over circuits using the D4 format.*




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