E1 carrier Bit Rate

E1 carrier comes with European digital transmission format devised by the ITU-TS (International Telecommunication Union Telecommunication Standardization Sector). This name is given by the Conference of European Postal and Telecommunication Administration (CEPT). Normally E1 carrier operates on European and some Asian countries. But E1 is equivalent standard of the T1 carrier system in North American. 

E2 through E5 are carriers in increasing multiples of the E1 format. But in the Industry only E1 and E3 versions are used. Actually E1 is transmitted as 32 time slots and E3 is capable of carrying 5120timeslots.There are two time slots for framing (For synchronization) and another one for signaling (For allocating call setup). E-carrier systems permanently assign capacity for a voice call for its whole period. Therefore this makes sure high call quality because the transmission arrives with the same short delay and capacity at all times.

Here are the data rates of multiple E carriers.

E0 - 64 kbit/s

E1 - 2.048 Mbit/s

E2 - 8.448 Mbit/s

E3 - 34.368 Mbit/s

E4 - 139.264 Mbit/s

E5 - 564.992 Mbit/s

In E1 carrier there are 32 channels in the frame. The bit rate can be calculated as follows;

Time taken for the whole frame              = 125μs

Number of bits in a time slot                   = 8 bits
Number of time slot in a frame               = 32
Therefore number of bits in a frame       = 32 x 8 bits

                                                                         265 bits

We know bit rate can be achieved as follows;

                        Bit rate of E1 carrier           = number of bits /time taken

                                                                          256 bits/125μs

                                                                          2.048 Mbit/s

PCM and TDM operation

Time division multiplexer is a digital processes that can be applied when the data rate capacity of the transmission medium is greater than the data rate required by the sending and receiving device.

·        After the Encoding each byte comes to the multiplexer, then multiplexer gather 30 bytes together (in every125 µs).

·        Then these 30 bytes (channels) are gathered as Frames. It is called E1 carrier.

·        Before multiplexing synchronizing bit and signaling bit is added to the frame.  Synchronizing bit is in the 1^st time slot and signaling bit is in the 16^th time slot.

·        Now this frame has 32 bytes and one by one goes to the destination end time division de-multiplexer.

·         De-multiplexer decomposes each frame by discarding the framing bits and extracting each character in turn.

Why we use Low Pass FIlter in PCM

If a low pass filter is not implemented prior to the sampling modulator, we have to face more problems, when the voice is passing through a copper cable.

 In sampling modulator we use a low pass filter to pass only low frequency of the voice signal. It compressed human voice bandwidth (20 kHz-20000 kHz) into small bandwidth of 0.3 kHz-3.4 kHz.

                      We know if want to transmit more distance, need a low bandwidth. Human voice bandwidth is a wider one, using that bandwidth can’t transmit that much of distance through the copper cable. It needs small bandwidth. That’s why we use low pass filter.

                      If it is not implemented prior to sampling modulator can’t transmit signal much distance through the copper cable.

                      And ccItt experimented on the range which most suitable for voice transmission and found out that 0-4kHz would be sufficient enough to transmit any message clearly.

What is PCM

Pulse code modulation (PCM) is used to transform an analogue wave form into digital that is compatible with a digital communication system, the following steps are taken.

Sampling- Digital signal transmission is done by having samples from analog signals. Sampling we obtain samples of an analogue signal within a certain time period. The analogue signal will sample at every sampling interval. According to the Nyquest theorem sampling frequency should be equal or greater than to the maximum frequency of the sound wave.

Quantization- Sampling gives a series of pulses with varying amplitude. Then the receiver can not identify the every finite value in the amplitude. So without sending the real sample we put the sample signals amplitude in to the recognized number of levels.  We ought to approximate the amplitude of the sample to the nearest quantization level. The heights between two levels are called a zone.

       

Encoding - Encoding means converting quantized levels into binary formats to suit for digital transmission. Encoding represent the sign, segment number and the sample

Compander in Pulse Code Modulation

Compander is a Signal processing technique which uses both compression and expansion to improve dynamic range and signal-to-noise ratio.

            Before the conversion of analog to digital in PCM, the compander boost up the low amplitudes of the voice.

In leaner quantization we give a low signal to distortion ratio and high ratio for high amplitudes. This is not useful, because if we get a low signal to distortion ratio, it let know the distortion is higher, compare to signal’s amplitude. Therefore we use compander for overcome that problem. What compander do is it only boost up the low amplitudes of the voice.

And also receiving end compander (expander) reduce the amplitude value of the boosted up signal.

But when we use nonlinear quantization there is a solution for this matter. It has lower gaps in quantized levels for low amplitudes and higher gaps in quantized levels for high amplitudes. Therefore we can achieve high signal to distortion ratio in both cases. And it is useful.

Analog over Digital

               The fundamental transmission concept is to transmit the minimum possible frequency or the bandwidth with maximum possible number of channels or the bit rate.

When we consider the analog transmission we can’t transmit much distance than digital transmission. Therefore we need to use small bandwidth. Using that small bandwidth can transmit much distance. Therefore we use modulation techniques to reduce the bandwidth. FDM multiplexing system is combining number of channels to convey over common transmission medium. Major advantage of analogue transmission is occupied band width is low When we use frequency modulation in low pass filter there are two, out come upper side bandwidth and lower side bandwidth. By choosing the lower side bandwidth we can reduce bandwidth of the signal.

             In digital transmission we can have more than solutions compare to analog.

Theoretically we know Nyquist Theorem, it tells us relationship between bandwidth and channel capacity.

                             C = 2Blog₂M

C- Channel capacity   ,    B – Bandwidth       ,    M – signal levels

If we want increase channel capacity for a value of bandwidth need to increase signal levels.

              In digital transmission we can use time division multiplexing (TDM).using this technique we can reduce bandwidth. In this TDM large numbers of channels are put into one line and transmit. . In the digital transmission can use the available bandwidth of the channel much more efficiently as compared to analogue system.