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.

The methods used for data communication over the PSTN

Normally the earlier PSTN is designed for carry voice, but after has been developed to carry data as well as voice.

There are many methods used for data communication over the PSTN. They are PSDN, ISDN, Dial up and  X DSL

       

PSDN (Public switched data Network) is runs over normal PSTN line. It is a Packet Switched data network that based on packet switching as opposed to circuit switching that used in PSTN. When we are connecting to the internet (WAN) we use normal PSTN twisted pair cabled. In this technology packets are routing according to the source and destination address.

 Integrated Services Digital Network (ISDN) turns the local loop into a TDM digital connection. It is purely digital network but runs on analog twister pair lines. Speed is one advantage ISDN has over telephone line connections. There are two ISDN implementations. They are,

 Basic Rate Interface (BRI)

 B channels – Bandwidth of 64kbps

D channels – Bandwidth of 16 kbps

          Designed as 2B + D

Primary rate Interface (PRI)

  B & D – Bandwidth of 64kbps

 The design varies according to the nation.

 Ex – North America & Japan 23B+D      

  

 

SDN Carries voice calls and data simultaneously on one line. It uses time division multiplexing (TDM) over two wire pairs. TDM Divides single channel into multiple channels.      

                         Dial-up connection also uses the PSTN for connect with WAN. It provides a low cost need based access. We can achieve bit rate up to 56kbps.In this method Modem is connected to a Telephone Line on the Customer End. Remote Access Server (RAS) is connected to Telephone Lines (33.6 Kbps connectivity) or E1/R2 Line (56 Kbps connectivity) on the Service Provider End. Then RAS provide dialing connectivity, authentication and metering. Disadvantage of this technology is achievable bandwidth depends on the line quality and the traffic of the network.

 

Digital Subscriber Line (DSL) technology is a broadband technology that uses existing twisted-pair telephone lines to transport high-bandwidth data to service subscribers. The term XDSL covers a number of similar yet competing forms of DSL technologies.DSL technology allows the local loop line to be used for normal telephone voice connection and an always-on connection for instant network connectivity. The two basic types of DSL technologies are asymmetric (ADSL) and symmetric (SDSL).  All forms of DSL service are categorized as ADSL or SDSL and there are several varieties of each type. Asymmetric service provides higher download or downstream bandwidth to the user than upload bandwidth. Symmetric service provides the same capacity in both directions.

Multiple DSL subscriber lines are multiplexed into a single, high capacity link by the use of a DSL Access Multiplexer (DSLAM) at the provider location. DSLAMs incorporate TDM technology to aggregate many subscriber lines into a less cumbersome single medium, generally T3/DS3 connection techniques to achieve data rates up to 8.192 Mbps.

                      

CISC Vs RISC Achitectures

CISC Vs RISC

          The performance of a CPU is highly dependent on its internal architecture. Mainly there are two architectures. They are,

CISC Architecture

        

          CISC stands for Complex Instruction Set Computing. Traditional theory states that CPUs can be made quicker by adding more and more complexity into the instructions of the instruction set. This architecture can execute several low level operations such as load from memory, an arithmetic operations and a memory store all in a single instruction.
         CISC is the more advanced architecture compare with RISC architecture. It can multiply two logical numbers in one step. In CISC Architecture no of bits per instruction is high. That means it can have high capacity. CISC is used in computer Micro processors & the devices which have high speed processing power.

RISC Architecture

              RISC stands for Reduced Instruction Set Computing. It states that the best performance can be achieved by reducing the time taken to execute any given instruction. Rather than have complex instructions that require many clock cycles (more on this later) to complete, RISC chips use very simple instructions that could be performed in fewer clock cycles. Performance can then be improved by making the cycles shorter.

             Compare to CISC, RISC haven’t that much of process power. It can’t multiply two logical numbers in one step. It uses add and shift function to do that.

Multiply function

CISC Architecture-    00010110 + 00000110 = 1000100     ; It can multiply it one step.

RISC Architecture-   00010110 + 00000110 =?  But here we can’t do it in one step because it doesn’t have a multiplying function.

Therefore it uses add & shift