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Digital Communications

Once there is a digital link between two sites, there is still the question of how to transfer data between them. Different practices are used for voice, digital and video communications. This document explains some of the methods used, as well as explains terms in digital multiplexing.

Analog to Digital conversion

The human voice is a continuous signal in the range 0-4 KHz. Digital communication on the other hand, is based on discrete bits (0 and 1). Therefore, there is a need for converting the human voice into a stream of bits and vice versa.

The analog to digital conversion is done by sampling the sound wave and denoting the level of the wave by a number which is transmitted over the digital link. The reverse process is done by creating a wave according to the received numbers. According to Nyquist law, the minimum number of such wave samples needed for complete reconstruction of the wave is twice the number of the maximum frequency of that wave.
This yields: 2*4K = 8K Samples per second.
The most common method for denoting the level of the wave is called PCM. These methods divide the level into 256 levels (8 bits).
Thus, if sampling 8K times a second, each sample in the range of 0-255 we need 8K * 8 = 64K bits per second per voice line.

Multiplexing and Synchronization

There are two problems that we need to solve:

  1. We would like to be able to transmit more than just 64Kb/s
  2. The receiving end should know where in the bit stream is the beginning of a new 8 bit number.

These two problems are addressed by multiplexing and the use of synchronization bits.

What is multiplexing ?

Clearly there is a need to transfer much more than a single channel between two sites. However, stretching a separate line for every channel is clearly not a good solution.
Multiplexing is a way of sending several (indeed - many) channels over a single line. This is done by using TDM - Time Division Multiplexing. Suppose we have 32 channels, each with a rate of 64Kbs, that we wish to transfer to the other end. The multiplexer takes from each of the 32 lines a single byte and sends them one after the other. After doing so, it takes the next byte from every channel, and so on.
Clearly, if we don't want bytes to get lost, the multiplexer must be able to send all the 32*8 bits from the 32 channels without the second byte of the first channel getting lost.
This implies that the output rate of the multiplexer should be at least 32*64Kbs or 2048 Kbs. This method is called Time Division Multiplexing (TDM) because the multiplexer took the 1/8000 sec needed for transferring a single byte of a single channel, and divided it between the 32 channels by increasing the rate so that each byte of a channel will take 1/(8000 * 32) sec to send.

Here is an example of multiplexing 3 channels of 64Kbs each:

This method could be further used for increasing the number of channels yet again from 32 channels to 4*32 channels and so on. Each increase is of course accompanied by a suitable increase in the bit rate of the line.

Well, we succeeded in sending 32 channels over a single line, but how will the receiving end (the demultiplexer) know which bit belongs to which channel?


Special bits in the bit stream are used for synchronization. These bits tell the demultiplexer where a new 32 byte group starts so it will know how to divide the following bits between the channels. No synchronization is needed for distinguishing between each of the 32 channels.
If we multiplex several 32 channels together, more synchronization bits are added for distinguishing between the different groups.



Digital data and Video

The upside for transmitting digital data or video is that no analog to digital conversion is needed. Instead, the bit stream in directly inserted into the multiplexer. Video, which needs a much higher bit rate than 64Kbs is usually inserted directly into the second level multiplexer, thus allowing a bit rate of 1.5-2 Mbs.


Obviously, standards have to be made if we would like equipment from different manufacturers to work with each other. Alas, there are more than one standard.

The most important ones are E1 which is used mainly in Europe and T1 which is used mainly in the US. and some far-eastern countries.
Although both standard start with a single channel rate being 64Kbs those channels are still incompatible because of the different ways by which the voice was digitized.


The first hierarchy of E1 is composed of 32 channels totaling 32*64Kbs = 2048 Kbs. Two of the channels are not used for transmitting data but for frame synchronization and signaling.


The first hierarchy of T1 is composed of 24 64Kbs channels plus 8 Kbs used for signaling totaling 24*64 + 8 = 1544 Kbs. The hierarchies are presented in the following table:


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