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CHAPTER 1 INTRODUCTION
Communication is the word coined for the practical application of scientific knowledge in the world. The advancement in communication cannot be justified unless it is used for leveraging the users purpose. Communication today imbibed the accomplishment of several tasks of varied complexity, almost in all aspects of life, the project here is meant for making the mobile communication easy and fast.
1.1 OBJECTIVE
With the increasing complexity of todays mobile networks and rising demands of their subscribers ,the telecommunication industry seeking for mobile communication experts able to provide project management under tight cost constraints and high quality requirements. The aim of the project is to benchmark the different network key performance indicators in different areas by conducting drive testing and compare the standards of different network performances in different areas considering the different KPIs. Network performance will be gauged with the help of drive test log values. RF network KPIs are calculated and comparative analysis will be done with TRAI benchmark values.
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Low cost Innovative Services Standard Interfaces 3.2.2 HISTORY OF MOBILE COMMUNICATION 1946 1960 1970 1979 1985 1986 1990 Appeared in St .Louis USA (By AT & T) at 150 MHz band FM 120 KHz BW 450 MHz Band FM 30 KHz BW
BELL LAB introduced Cellular Principle Advanced Mobile Phone System in US Total Access Communication System (TACs in UK) Nordic Mobile Telephony Systems (NMT) Digital Systems
Multi Standard
Multi Standard
Unified
Standard
Only
mobile
Mostly
for
voice Voice
and
data
voice services
services & data delivery (breaking data barrier )possible mainly data Terrestrial
Terrestrial
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Cellular systems are described in multiple generations, with third and fourth generation (3G and 4G) systems just emerging. 1G systems: These are the analog systems such as AMPS that grew rapidly in the 1980s and are still available today. Many metropolitan areas have a mix of 1G and 2G systems, as well as emerging 3G systems. The systems use frequency division multiplexing (FDMA) to divide the bandwidth into specific frequencies that are assigned to individual calls. 2G systems: These second-generation systems are digital, and use either TDMA (Time Division Multiple Access) or CDMA (Code Division Multiple Access) access methods. The European GSM (Global System for Mobile communications) is a 2G digital system with its own TDMA access methods. The 2G digital services began appearing in the late 1980s, providing expanded capacity and unique services such as caller ID, call forwarding, and short messaging. A critical feature was seamless roaming, which lets subscribers move across provider boundaries. 3G systems: 3G has become an umbrella term to describe cellular data communications with a target data rate of 2 M bits/sec. The ITU originally attempted to define 3G in its IMT-2000 (International Mobile Communications-2000) specification, which specified global wireless frequency ranges, data rates, and availability dates. However, a global standard was difficult to implement due to different frequency allocations around the world and conflicting input. So, three operating modes were specified. In general, a 3G device will be a personal, mobile, multimedia communications device that supports speech, color pictures, and video, and various kinds of information content. There is some doubt that 3G systems will ever be able to deliver the bandwidth to support these features because bandwidth is shared. However, 3G systems will certainly support more phone calls per cell. 4G Systems: On the horizon are 4G systems that may become available even before 3G matures (3G is a confusing mix of standards). While 3G is important in boosting the number of wireless calls, 4G will offer true high-speed data services. 4G data rates will be in the 2-Mbit/sec to 156Mbit/sec range, and possibly higher. 4G will also fully support IP. High data rates are due to advances in signal processors, new modulation techniques, and smart antennas that can focus signals directly at users. OFDM (orthogonal frequency division multiplexing) is one scheme that can provide very high wireless data rates. OFDM is described under its own heading.
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1984
Establishment of three working parties (WP1-3) to define and describe the services offered in a GSM PLMN (GSM Public Land Mobile Network) the radio interface, transmission, signaling protocols, interfaces and network architecture.
1986
A so called permanent nucleus is established to continuously coordinate the work, which is intensely supported by industry delegates.
1987
Initial memorandum of understanding (MOU) signed by network operator organizations (representing 12 countries) with major objectives as: * Coordinating the introduction of the standard and time scales. * Planning of service introduction * Routing, billing, and tariff coordination.
1988/89 To 1991/92
With the establishment of the European telecommunication Standards Institute (ETSI), the specification work was mooted to This international body. GSM becomes a technical committee within ETSI and splits up to into GSM groups 1-4, later called Special Mobile Groups (SMG) 1-4, which are technical sub Committees. GSM finally stands for Global system for Mobile Communications
1990
The GSM specifications for 900 MHz band are also applied to a Digital cellular system on the 1800 MHz band (DCS1800), a PCN application initiated in the United Kingdom.
The GSM Recommendations comprise more than 130 single documents. Official commercial launch of GSM service in Europe. The GSM- MOU has 62 members (signatories) in 39 countries worldwide. The end of 1993 shows one million subscribers to GSM networks, however more than 80% of them is to be found in Germany alone.
1993
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The features and benefits expected in the new system were Superior speech quality Low terminal, operational, and service costs A high level of security (confidentiality and fraud prevention) International roaming Support of low terminal hand portable terminals A variety of new services and network facilities.
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and dispersion .Fading is due to the interference of multiple signals with random relative phase that causes variations in the amplitude of the received signal. This will increase the error rate in digital systems, since errors will occur when the signal-to-noise ratio drops below certain
Figure 3.4 Multipath Radio environments. threshold. Dispersion is due to differences in the delay of the various paths, which disperses transmitted pulses in time. If the variation of the delay is comparable with the symbol period, delayed signals from an earlier symbol may interfere with the next symbol; causing Inter-symbol interference (ISI).The counter measures for fading include diversity reception and equalization. (c) Mobility management The principal characteristic of mobile networks, which distinguishes them from conventional fixed networks, is that the identity of calling and called subscribers is not associated with a fixed geographical location. The subscribers establish a wireless connection with the nearest base station, and can make or receive calls as they roam. Mobility management is concerned with how the network supports this function. When a call is made to mobile customer, the network must be able to locate the mobile customer. Network attachment process which includes a location updating process is the answer for the mobility management. In the location update process, the network databases are updated dynamically, so that the mobile can be reached to offer the services. If this process is not done efficiently, it will result in poor call management and network congestion.
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(d) Services International roaming shall be provided. Advanced PSTN services should be provided consistent with ISDN services albeit at limited bit rates only. Encryption should be used to improve security for both the operators and the customers. (e) Network aspects ITU identification and numbering plans should be used an international signaling system should be utilized. There should be a choice of charging structure and rates. No modification shall be required to the PSTN due to its interconnection to GSM signaling and control information should be protected. (f) Cost The system parameters should be chosen to limit costs, particularly mobiles and handsets. In a competitive environment, cost is the deciding factor for the survival of an operator.
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procedures. The mechanisms include modulation, power control, coding, timing, and host of other details that manage the establishment and maintenance of the channel. The radio channel has to be fully exploited for maximum capacities and optimum quality of service. Band width is a scarce natural resource. The bandwidth has to be managed for maximum capacity of the system and interference free communications. The spectrum availability for an operator is very limited. The uplink or down link spectrum is only 25 MHz, Out of this 25 MHz, 124 carriers of each 200 KHz are generated. These carriers are to be shared amongst different operators. And as a result each operator gets only a few tens of carriers; making spectrum management a challenging area. The following figure shows the radio connectivity between the mobile equipment and the Radio transmitter/receiver. Radio interface
MOBILE SWITCH
RADIO CONTROLLER
RADIO TRANSCEIVER
MOBILE
Figure 4.1 Radio Communication between mobile and Tx/Rx For effective management of bandwidth, for conservation of spectrum and quality of radio link; the following access techniques are implemented on the radio interface.
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Cellular structures and Frequency Reuse Multiple access Technologies Voice coding technologies Bandwidth effective Modulation scheme.
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(a) Cellular System Characteristics The distinguishing features of digital cellular systems compared to other mobile radio systems are: Small cells small coverage radii (on the
A cellular system uses many base stations with relatively order of a 100 m to 30 km). Clusters and Frequency reuse
The spectrum allocated for a cellular network is limited. As a result there is a limit to the number of channels or frequencies that can be used. A group of cells is called a cluster. All the frequencies are used in a cluster and no frequency is reused within the cluster. And the total set of frequencies is repeated in the adjacent cluster. Like that the total service area, i.e. may be a country or a continent, can be served with a small group of frequencies. Frequency reuse is possible because the signal fades over the distance and hence it can be reused .For this reason each frequency is used simultaneously by multiple base-mobile pairs; located at geographically distant cells. This frequency reuse allows a much higher subscriber density per MHz of spectrum than other systems. System capacity can be further increased by reducing the cell size (the coverage area of a single base station), down to radii as small as 200m. Small, battery-powered handsets In addition to supporting much higher densities than previous systems, this
approach enables the use of small, battery-powered handsets with a radio frequency that is lower than the large mobile units used in earlier systems. Performance of handovers In cellular systems, continuous coverage is achieved by executing a handover (the seamless transfer of the call from one base station to another) as the mobile unit crosses cell boundaries. This requires the mobile to change frequencies under control of the cellular network.
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(b) Co-channel cells and interference Radio channels can be reused provided the separation between cells containing the same channel set is far enough apart so that co-channel interference can be kept below acceptable levels most of the time. Cells using the same channel set are called Co-channel cells. Co-channel cells interfere with each other and quality is affected. Within the service area (PLMN), specific channel sets are reused at a different location (another cell). In the example, there are 7 channel sets: A through G. Neighboring cells are not allowed to use the same frequencies. For this reason all channel sets are used in a cluster of neighboring cells. As there are 7 channel sets, the PLMN can be divided into clusters of 7 cells each. Co-channel interference Frequencies can be reused throughout a service area because radio signals typically attenuate with distance to the base station (or mobile station). When the distance between cells using the same frequencies becomes too small, co-channel Interference might occur and lead to service interruption or unacceptable quality of service. As long as the ratio Frequency reuse distance = D Cell radius is greater than some specified value, the ratio Received radio carrier power Received interferer radio carrier power = C I R
will be greater than some given amount for small as well as large cell sizes; when all signals are transmitted at the same power level. The average attenuation of radio signals with distance in most cellular systems is a reduction to about 1/16 of the received power for every doubling of distance (1/10000 per decade). The frequency reuse distance known as separation distance is also known as the signal-tonoise ratio. The figure on the opposite page shows the situation. At the base station, both signals from subscribers within the cell covered by this base station and signals from subscribers covered by other cells are received. Interference is caused by cells using the same channel set. The ratio D/R needs to be large enough in order for the base station to be able to cope with the interference. A co-channel interference factor Q is defined
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As Q=D/R = 3K where D is Frequency reuse distance R is the cell radius and K is the reuse factor or the number of cells in a cluster.
Capacity / performance of trade-offs When engineering a cellular network, the most important trade-off to make is the one
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The performance of a cellular network can be expressed in quality of service. That is the value of Q shall be higher to achieve an acceptable quality of service. This means a low (cochannel) interference level in the network.
The relationship between the reuse factor K and the network performance are: if K increases, then the co-channel interference decreases, and so the performance increases (note that there is a fixed relationship between K and ratio D/R). Relationship between K and Cell Capacity The other key relationship in cellular networks is the one between the reuse factor K and call capacity. First of all, call capacity depends on the number of available channels. In GSM, a limited number of frequencies is available (for GSM: 124 frequencies, and for GSM-1800: 374 frequencies). The frequencies are grouped into frequency sets. If K increases, the number of frequencies per set (and so per cell) decreases, and so the call capacity per cell. The value of K in GSM cellular networks varies between 4 and 21. Note that in real networks, K is not a constant within the whole PLMN area, but varies depending on the traffic capacity needed in certain regions. Typically, K is high in urban regions and low in rural regions. If K increases, then performance increases If K increases, then call capacity decreases per cell The number of sites to cover a given area with a given high traffic density, and hence the cost of the infrastructure, is determined directly by the reuse factor and the number of traffic channels that can be extracted from the available spectrum. These two factors are compounded in what is called spectral efficiency of the system. Not all systems allow the same performance in this domain: they depend in particular on the robustness of the radio transmission scheme against interference, but also on the use of a number of technical tricks, such as reducing transmission during the silences of a speech communication. The spectral efficiency, together with the constraints on the cell size, determines also the possible compromises between the capacity and the cost of the infrastructure. All this explains the importance given to spectral efficiency.
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bandwidth with some reliability. In any cellular system or cellular technology, it is necessary to have a scheme that enables several multiple users to gain access to it and use it simultaneously. As cellular technology has progressed different multiple access schemes have been used. They form the very core of the way in which the radio technology of the cellular system works. A mix of Frequency Division Multiple Access (FDMA) and Time Division Multiple Access (TDMA), combined with frequency hopping, has been adopted as the multiple access schemes for GSM. GSM chose a combination of TDMA/FDMA as its method. The FDMA part involves the division by frequency of the total 25 MHz bandwidth into 124 carrier frequencies of 200 kHz bandwidth. One or more carrier frequencies are then assigned to each BS. Each of these carrier frequencies is then divided in time, using a TDMA scheme, into eight time slots. One time slot is used for transmission by the mobile and one for reception. They are separated in time so that the mobile unit does not receive and transmit at the same time.
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ECE transmitting carrier .The digital information generated by the system or the network is to be imparted to the radio carrier by suitable digital modulation technique. If the amplitude of a carrier is shifted with binary information, it is said ASK is employed, wherein the amplitude of the carrier is switched between their full-on and full-off conditions. If the carrier frequency is shifted with the binary information, this is equivalent to shifting between two or more carriers of different frequencies. This is FSK and is widely used in analog cellular systems for signaling functions. There is no limit to the number of carrier frequencies that can be shifted, but the use of two frequencies, quite close together, is the universal implementation of FSK. As with FSK ,the shift between various carriers differing from each other only in their relative phase(PSK).There are many varieties of PSK ,and each is broadly distinguished from the others by the number of allowed phases . 4.4.1 GAUSSIAN MINIMUM SHIFT KEYING (GMSK) The modulation specified for GSM is GMSK with BT=0.3 and rate 270 5/6 k bauds. GMSK is a type of constant envelope FSK, where the frequency modulation is a result of a carefully contrived phase modulation .The most important feature of GMSK is that it is a constant envelope variety of modulation. This means there is a distinct lack of AM in the carrier with a constant limiting of the occupied bandwidth. The constant amplitude of the GMSK signal makes it suitable for use with high efficiency amplifiers. An easy way to understand the GMSK signal is to first investigate its precursor, MinimumShift Keying (MSK).The following figure indicates the steps in generating an MSK signal. How the data is treated in GMSK is explained below: The waveforms are all aligned together in phase. Little scales are placed are placed in the figure to help make the phase relationships between the waveforms clearer. 10 bits of the data stream {1101011000} is considered for analysis. The data stream is divided into odd and even bit streams:(odd bits and even bits).In creating odd bits and even bits ,each alternate odd and even bit in data is hold for two bit times. Staggering odd bits and even bits already helps to create a waveform with minimal AM. For convenience odd bits and even bits are made to
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The fourth and fifth wave forms in the following figure are the high freq and the low freq versions, respectively, of the carrier. Since MSK is a form of FSK, finally
modulated carrier needs two diff. Frequency components (low and high).the MSK signal is created by shifting between these two frequencies. The MSK signal is created starting with bit number 2, with the help of the truth table given below along with the waveforms. At any instant the odd and even bit values are taken from the table and follow the rules as given in the truth table to obtain the MSK waveform at that instant. Either the high or the low frequency versions of the carrier is picked corresponding to the instant under consideration and also according to the sense instructions(+or-) the wave form is to be turned up or down.
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Figure 4.4 GMSK wave forms Smooth phase transitions can be noticed, as the MSK waveform changes its frequency one from the other. These high and low frequencies shall be as close together as possible in the frequency domain. To make a GMSK signal from an MSK signal ,the stretched data waveforms (each135.4 kbps) have to be filtered with a Gaussian filter of an appropriate bandwidth defined by the BT product(Bandwidth*Time).In GSM case ,BT is 0.3,which makes B=81.3 kHz when T is 3.7 micro sec (T=1/270.833).
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ECE too high to be economically transmitted over the radio interface; considering the Bandwidth scarcity. Band width has to be shared by number of users for costing advantages. The speech coder will have to do something to significantly reduce this rate by extracting irrelevant components in the data stream. The speech coder has to search for excess baggage we can safely remove from the bit stream scheduled for transport over the radio path. GSM uses to processes to strip redundant fat from the data representing voice traffic. The compression algorithm used in GSM is a procedure called RPE-LTP.
4.5.2 REGULAR PULSE EXCITATION AND LONG TERM PREDICTION (RPELTP) Every 20ms, 160 sampled values from the ADC are taken and stored in an intermediate memory. An analysis of a set of data samples produces eight filter coefficients and an excitation signal for a time-invariant digital filter. This filter can be regarded as a digital imitation of the human vocal tract, where the finer coefficients represent vocal modifiers(e.g., teeth, tongue, pharynx)and the excitation signal represents the sound(e.g., pitch , loudness) or the absence of sound that we pass through the vocal tract(filter). A correct setting of filter coefficients and an appropriate excitation signal yields a sound typical of the human voice. The procedure, so far, has not performed any data reductions. The reductions come in further steps, which take advantage of certain attributes of the human ear and vocal tract .The 160 samples, transformed into filter coefficients, are divided into four blocks of 40 samples each. Each block represents a 5-ms period of voice. These blocks are sorted into four sequences. Where each sequence contains very forth sample from the original 160 samples. Sequence number 1 contains samples 1, 5, 9, 13., 37, sequence number 2 contains samples 2, 6, 10, 14, .38, Sequence number 3 contains samples3, 7, 11, 15,39, and Sequence number 4 contains samples 4, 8, 12, 1640. The first reduction in data comes when the speech encoder selects the sequence with the most energy. This linear predictive coding (LPC) and regular pulse excitation (RPE) analysis has a very short memory of approximately 1ms. A more long-term consideration of neighboring (or adjacent) blocks in time is not performed here, There are numerous correlations in the human voice, especially in long vowels such as the in car, where the same sound recurs in succeeding 523
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ECE ms samples. Taking the similarity of sounds between adjacent samples (Adjacent 5-ms blocks) into account can significantly reduce the amount of data required to describe the human voice. This second reduction task is performed by a LTP Function. 4.5.3 LONG-TERM PREDICTION ANALYSIS (LTP) The LTP function accepts a sequence selected by the LPC/RPE analysis. Upon accepting sequence, it then looks among all the previous sequences passed to it (which will reside in another intermediate memory for 15ms) for the earlier sequence that has the highest correlation to ( bears the greatest resemblance to ) the current sequence. It can be said that the LTP function
looks for the one sequence from among those already received that is most similar to the sequence just received from the LPC/RPE. Now it is only necessary to transmit a value representing the difference between the two sequences, along with a pointer to tell the receiver on the other end of the radio channel, which sequence it should select among its recently received sequences for comparison. The receiver knows which differential values it has to apply to which sequences. The transmission of the whole sequence is not necessary, only the difference between sequences, This further reduces the data on the channel. The speech coder issues a block of 260bits (a speech frame) once every 20ms. This corresponds to net data rate of 13kbps, a data reduction of a factor of eight. Speech transcoding is a task that requires a large number of calculations at high speeds. It is, therefore, an ideal application for digital signal processing (DSP) techniques.
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Figure 5.1 Architecture of GSM Network. In order to ensure that network operators will have several sources of cellular infrastructure equipment, GSM decided to specify not only the air interface, but also the main interfaces that identify different parts. There are three dominant interfaces, namely, an interface between MSC and the base Transceiver Station (BTS), and an Um interface between the BTS and MS.
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The GSM service is the total area served by the combination of all member countries where a mobile can be serviced. The next level is the PLMN service area. There can be several within a country, based on its size. The links between a GSM/PLMN network and other PSTN, ISDN, or PLMN network will be on the level of international or national transit exchange. All incoming calls for a GSM/PLMN network will be routed to a gateway MSC. A gateway MSC works as an incoming transit exchange for the GSM/PLMN. In a GSM/PLMN network, all mobile-terminated calls will be routed to a gateway MSC. Call connections between PLMNs, or to fixed networks, must be routed through certain designated MSCs called a gateway MSC. The gateway MSC contains the interworking functions to make these connections. They also route incoming calls to the proper MSC within the network. The next level of division is the MSC/VLR service area. In one PLMN there can be several MSC/VLR service areas. MSC/VLR is a role controller of calls within its jurisdiction. In order to route a call to a mobile subscriber, the path through links to the MSC in the MSC area where the subscriber is currently located. The mobile location can be uniquely identified since the MS is registered in a VLR, which is generally associated with an MSC. The next division level is that of the LAs within a MSC/VLR combination. There are several LAs within one MSC/VLR combination. A LA is a part of the MSC/VLR service area in which a MS may move freely without updating location information to the MSC/VLR
exchange that control the LA. Within a LA a paging message is broadcast in order to find the called mobile subscriber. The LA can be identified by the system using the Location Area Identity (LAI). The LA is used by the GSM system to search for a subscriber in an active state. Lastly, a LA is divided into many cells. A cell is an identity served by one BTS. The MS distinguishes between cells using the Base Station Identification code (BSIC) that the cell site broadcast over the air.
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ECE connection of external terminals, such as the interface for a personnel computer or fax. The MS includes mobile equipment and a subscriber identity module (SIM).MS does not need to be personally assigned to one subscriber. The SIM is a subscribe module which stores all the subscriber-related information. When a subscribers SIM is inserted into the ME of an MS that MS belongs to the subscriber, and the call is delivered to that Ms. The ME is not associated with a called number-it is linked to the SIM. In this case, any ME can be used by a subscriber when the SIM is inserted in the MS. SIM is needed in order to access the services provided by the GSM PLMN. MS can be installed in Vehicles or can be portable or handheld stations. The MS may include provisions for data communication as well as voice. A mobile transmits and receives message to and from the GSM system over the air interface to establish and continue connections through the system. Different type of MSs can provide different type of data interfaces. To provide a common model for describing these different MS configuration, reference configuration for MS, similar to those defined for ISDN land stations, has been defined. Each MS is identified by an IMEI that is permanently stored in the mobile unit. Upon request, the MS sends this number over the signaling channel to the MSC. The IMEI can be used to identify mobile units that are reported stolen or operating incorrectly. Just as the IMEI identities the mobile equipment, other numbers are used to identify the mobile subscriber. Different subscriber identities are used in different phases of call setup. The Mobile Subscriber ISDN Number (MSISDN) is the number that the calling party dials in order to reach the subscriber. It is used by the land network to route calls toward an appropriate MSC. The international mobile subscribe identity (IMSI) is the primary function of the subscriber within the mobile network and is permanently assigned to him. The GSM system can also assign a Temporary Mobile Subscriber Identity (TMSI) to identity a mobile. This number can be periodically changed by the system and protect the subscriber from being identified by those attempting to monitor the radio channel. 5.3.1 FUNCTIONS OF MS The primary functions of MS are to transmit and receive voice and data over the air interface of the GSM system. MS performs the signal processing function of digitizing, encoding, error protecting, encrypting, and modulating the transmitted signals. It also performs
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the inverse functions on the received signals from the BS. In order to transmit voice and data signals, the mobile must be in synchronization with the system so that the messages are the transmitted and received by the mobile at the correct instant. To achieve this, the MS automatically tunes and synchronizes to the frequency and TDMA timeslot specified by the BSC. This message is received over a dedicated timeslot several times within a multiform period of 51 frames. We shall discuss the details of this in the next chapter. The exact synchronization will also include adjusting the timing advance to compensate for varying distance of the mobile from the BTS. The MS monitors the power level and signal quality, determined by the BER for known receiver bit sequences, from both its current BTS and up to six surrounding BTSs. This data is received on the downlink broadcast control channel. The MS determines and send to the current BTS a list of the six best-received BTS signals. The measurement results from MS on downlink quality and surrounding BTS signal levels are sent to BSC and processed within the BSC. The system then uses this list for best cell handover decisions. MS keeps the GSM network informed of its location during both national and international roaming, even when it is inactive. This enables the system to page in its present LA. The MS includes an equalizer that compensates for multi-path distortion on the received signal. This reduces inter-symbol interference that would otherwise degrade the BER. Finally, the MS can store and display short received alphanumeric messages on the liquid crystal display (LCD) that is used to show call dialing and status information. These messages are limited to 160 characters in length.
Power Levels These are five different categories of mobile telephone units specified by the European
GSM system: 20W, 8W, 5W, 2W, and 0.8W. These correspond to 43-dBm, 39-dBm, and 37dBm, 33-dBm, and 29-dBm power levels. The 20-W and 8-W units (peak power) are either for vehicle-mounted or portable station use. The MS power is adjustable in 2-dB steps from its nominal value down to 20mW (13 dBm). This is done automatically under remote control from the BTS, which monitors the
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ECE received power and adjusts the MS transmitter to the minimum power setting necessary for reliable transmission. 5.3.2 SIM CARD As described in the first chapter, GSM subscribers are provided with a SIM card with its unique identification at the very beginning of the service. By divorcing the subscriber ID from the equipment ID, the subscriber may never own the GSM mobile equipment set. The subscriber is identified in the system when he inserts the SIM card in the mobile equipment. This provides an enormous amount of flexibility to the subscribers since they can now use any GSM-specified mobile equipment. Thus with a SIM card the idea of Personalize the equipment currently in use and the respective information used by the network (location information) needs to be updated. The smart card SIM is portable between Mobile Equipment (ME) units. The user only needs to take his smart card on a trip. He can then rent a ME unit at the destination, even in another country, and insert his own SIM. Any calls he makes will be charged to his home GSM account. Also, the GSM system will be able to reach him at the ME unit he is currently using. The SIM is a removable SC, the size of a credit card, and contains an integrated circuit chip with a microprocessor, random access memory (RAM), and read only memory (ROM). It is inserted in the MS unit by the subscriber when he or she wants to use the MS to make or receive a call. As stated, a SIM also comes in a modular from that can be mounted in the subscribers equipment. When a mobile subscriber wants to use the system, he or she mounts their SIM card and provide their Personal Identification Number (PIN), which is compared with a PIN stored within the SIM. If the user enters three incorrect PIN codes, the SIM is disabled. The PIN can also be permanently bypassed by the service provider if requested by the subscriber. Disabling the PIN code simplifies the call setup but reduces the protection of the users account in the event of a stolen SIM.
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ECE authorized subscriber and is the only absolute identity that a subscriber has within the GSM system. The IMSI consists of the MCC followed by the NMSI and shall not exceed 15 digits. 5.4.2 TEMPORARY MOBILE SUBSCRIBER IDENTITY (TMSI) A TMSI is a MSC-VLR specific alias that is designed to maintain user confidentiality. It is assigned only after successful subscriber authentication. The correlation of a TMSI to an IMSI only occurs during a mobile subscribers initial transaction with an MSC (for example, location updating). Under certain condition (such as traffic system disruption and malfunctioning of the system), the MSC can direct individual TMSIs to provide the MSC with their IMSI. 5.4.3 MOBILE STATION ISDN NUMBER (MSISDN) The MS international number must be dialed after the international prefix in order to obtain a mobile subscriber in another country. The MSISDN numbers is composed of the country code (CC) followed by the National Significant Number (NSN), which shall not exceed 15 digits. 5.4.4 MOBILE STATION ROAMING NUMBER (MSRN) The MSRN is allocated on temporary basis when the MS roams into another numbering area. The MSRN number is used by the HLR for rerouting calls to the MS. It is assigned upon demand by the HLR on a per-call basis. The MSRN for PSTN/ISDN routing shall have the same structure as international ISDN numbers in the area in which the MSRN is allocated. The HLR knows in what MSC/VLR service area the subscriber is located. At the reception of the MSRN, HLR sends it to the GMSC, which can now route the call to the MSC/VLR exchange where the called subscriber is currently registered.
5.4.5 INTERNATIONAL MOBILE EQUIPMENT IDENTITY (IMEI) The IMEI is the unique identity of the equipment used by a subscriber by each PLMN and is used to determine authorized (white), unauthorized (black), and malfunctioning (gray) GSM hardware. In conjunction with the IMSI, it is used to ensure that only authorized users are granted access to the system. An IMEI is never sent in cipher mode by MS.
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ECE The BSS is a set of BS equipment consisting of a Radio transmitter/receiver called BTS
(Base Transceiver Station)and a controller called BSC (Base Station Controller).The BSS is viewed by the MSC through a single A interface as being the entity responsible for communicating with MSs in a certain area. The radio equipment of a BSS may be composed of one or more cells. A BSS may consist of one or more BTS. The interface between BSC and BTS is designed as an A-bis interface. The BSS includes two types of machines: the BTS in contact with the MSs through the radio interface and the BSC, the latter being in contact with the MSC. The function split is basically between transmission equipment, the BTS, and managing equipment at the BSC. A BTS compares radio transmission and reception devices, up to and including the antennas, and also all the signal processing specific to the radio interface. A single transceiver within BTS supports eight basic radio channels of the same TDM frame. A BSC is a network component in the PLMN that function for control of one or more BTS. It is a functional entity that handles common control functions within a BTS. The performance of a cellular network can be expressed in quality of service. That is the value of Q shall be higher to achieve an acceptable quality of service. This means a low (cochannel) interference level in the network.
The relationship between the reuse factor K and the network performance are: if K increases, then the co-channel interference decreases, and so the performance increases (note that there is a fixed relationship between K and ratio D/R). Relationship between K and Performance The performance of a cellular network can be expressed in quality of service. That is the value of Q shall be higher to achieve an acceptable quality of service. This means a low (cochannel) interference level in the network.
The relationship between the reuse factor K and the network performance are: if K increases, then the co-channel interference decreases, and so the performance increases (note that there is a fixed relationship between K and ratio D/R).
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Relationship between K and Cell Capacity The other key relationship in cellular networks is the one between the reuse factor K and call capacity. First of all, call capacity depends on the number of available channels. In GSM, a limited number of frequencies is available (for GSM: 124 frequencies, and for GSM-1800: 374 frequencies). The frequencies are grouped into frequency sets. If K increases, the number of frequencies per set (and so per cell) decreases, and so the call capacity per cell. The value of K in GSM cellular networks varies between 4 and 21. Note that in real networks, K is not a constant within the whole PLMN area, but varies depending on the traffic capacity needed in certain regions. Typically, K is high in urban regions and low in rural regions. If K increases, then performance increases If K increases, then call capacity decreases per cell The number of sites to cover a given area with a given high traffic density, and hence the cost of the infrastructure, is determined directly by the reuse factor and the number of traffic channels that can be extracted from the available spectrum. These two factors are compounded in what is called spectral efficiency of the system. Not all systems allow the same performance in this domain: they depend in particular on the robustness of the radio transmission scheme against interference, but also on the use of a number of technical tricks, such as reducing transmission during the silences of a speech communication. The spectral efficiency, together with the constraints on the cell size, determines also the possible compromises between the capacity and the cost of the infrastructure. All this explains the importance given to spectral efficiency.
bandwidth with some reliability. In any cellular system or cellular technology, it is necessary to have a scheme that enables
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ECE several multiple users to gain access to it and use it simultaneously. As cellular technology has progressed different multiple access schemes have been used. They form the very core of the way in which the radio technology of the cellular system works. A mix of Frequency Division Multiple Access (FDMA) and Time Division Multiple Access (TDMA), combined with frequency hopping, has been adopted as the multiple access schemes for GSM. GSM chose a combination of TDMA/FDMA as its method. The FDMA part involves the division by frequency of the total 25 MHz bandwidth into 124 carrier frequencies of 200 kHz bandwidth. One or more carrier frequencies are then assigned to each BS. Each of these carrier frequencies is then divided in time, using a TDMA scheme, into eight time slots. One time slot is used for transmission by the mobile and one for reception. They are separated in time so that the mobile unit does not receive and transmit at the same time.
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ECE systems for signaling functions. There is no limit to the number of carrier frequencies that can be shifted, but the use of two frequencies, quite close together, is the universal implementation of FSK. As with FSK ,the shift between various carriers differing from each other only in their relative phase(PSK).There are many varieties of PSK ,and each is broadly distinguished from the others by the number of allowed phases . 4.4.1 GAUSSIAN MINIMUM SHIFT KEYING (GMSK) The modulation specified for GSM is GMSK with BT=0.3 and rate 270 5/6 k bauds. GMSK is a type of constant envelope FSK, where the frequency modulation is a result of a carefully contrived phase modulation .The most important feature of GMSK is that it is a constant envelope variety of modulation. This means there is a distinct lack of AM in the carrier with a constant limiting of the occupied bandwidth. The constant amplitude of the GMSK signal makes it suitable for use with high efficiency amplifiers. An easy way to understand the GMSK signal is to first investigate its precursor, MinimumShift Keying (MSK).The following figure indicates the steps in generating an MSK signal. How the data is treated in GMSK is explained below: The waveforms are all aligned together in phase. Little scales are placed are placed in the figure to help make the phase relationships between the waveforms clearer. 10 bits of the data stream {1101011000} is considered for analysis. The data stream is divided into odd and even bit streams:(odd bits and even bits).In creating odd bits and even bits ,each alternate odd and even bit in data is hold for two bit times. Staggering odd bits and even bits already helps to create a waveform with minimal AM. For convenience odd bits and even bits are made to take the values 1or1. In GSM case, if the data rate (in waveform data) is 270.833 kbps, then the staggered odd bits and even bits will have half the rate135.4 kbps. The fourth and fifth wave forms in the following figure are the high freq and the low freq versions, respectively, of the carrier. Since MSK is a form of FSK, finally
modulated carrier needs two diff. Frequency components (low and high).the MSK
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signal is created by shifting between these two frequencies. The MSK signal is created starting with bit number 2, with the help of the truth table given below along with the waveforms. At any instant the odd and even bit values are taken from the table and follow the rules as given in the truth table to obtain the MSK waveform at that instant. Either the high or the low frequency versions of the carrier is picked corresponding to the instant under consideration and also according to the sense instructions(+or-) the wave form is to be turned up or down.
Figure 4.4 GMSK wave forms Smooth phase transitions can be noticed, as the MSK waveform changes its frequency one from the other. These high and low frequencies shall be as close together as possible in the
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frequency domain. To make a GMSK signal from an MSK signal ,the stretched data waveforms (each135.4 kbps) have to be filtered with a Gaussian filter of an appropriate bandwidth defined by the BT product(Bandwidth*Time).In GSM case ,BT is 0.3,which makes B=81.3 kHz when T is 3.7 micro sec (T=1/270.833).
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4.5.2 REGULAR PULSE EXCITATION AND LONG TERM PREDICTION (RPELTP) Every 20ms, 160 sampled values from the ADC are taken and stored in an intermediate memory. An analysis of a set of data samples produces eight filter coefficients and an excitation signal for a time-invariant digital filter. This filter can be regarded as a digital imitation of the human vocal tract, where the finer coefficients represent vocal modifiers(e.g., teeth, tongue, pharynx)and the excitation signal represents the sound(e.g., pitch , loudness) or the absence of sound that we pass through the vocal tract(filter). A correct setting of filter coefficients and an appropriate excitation signal yields a sound typical of the human voice. The procedure, so far, has not performed any data reductions. The reductions come in further steps, which take advantage of certain attributes of the human ear and vocal tract .The 160 samples, transformed into filter coefficients, are divided into four blocks of 40 samples each. Each block represents a 5-ms period of voice. These blocks are sorted into four sequences. Where each sequence contains very forth sample from the original 160 samples. Sequence number 1 contains samples 1, 5, 9, 13., 37, sequence number 2 contains samples 2, 6, 10, 14, .38, Sequence number 3 contains samples3, 7, 11, 15,39, and Sequence number 4 contains samples 4, 8, 12, 1640. The first reduction in data comes when the speech encoder selects the sequence with the most energy. This linear predictive coding (LPC) and regular pulse excitation (RPE) analysis has a very short memory of approximately 1ms. A more long-term consideration of neighboring (or adjacent) blocks in time is not performed here, There are numerous correlations in the human voice, especially in long vowels such as the in car, where the same sound recurs in succeeding 5ms samples. Taking the similarity of sounds between adjacent samples (Adjacent 5-ms blocks) into account can significantly reduce the amount of data required to describe the human voice. This second reduction task is performed by a LTP Function. 4.5.3 LONG-TERM PREDICTION ANALYSIS (LTP) The LTP function accepts a sequence selected by the LPC/RPE analysis. Upon accepting sequence, it then looks among all the previous sequences passed to it (which will reside in another intermediate memory for 15ms) for the earlier sequence that has the highest correlation to ( bears the greatest resemblance to ) the current sequence. It can be said that the LTP function
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looks for the one sequence from among those already received that is most similar to the sequence just received from the LPC/RPE. Now it is only necessary to transmit a value representing the difference between the two sequences, along with a pointer to tell the receiver on the other end of the radio channel, which sequence it should select among its recently received sequences for comparison. The receiver knows which differential values it has to apply to which sequences. The transmission of the whole sequence is not necessary, only the difference between sequences, This further reduces the data on the channel. The speech coder issues a block of 260bits (a speech frame) once every 20ms. This corresponds to net data rate of 13kbps, a data reduction of a factor of eight. Speech transcoding is a task that requires a large number of calculations at high speeds. It is, therefore, an ideal application for digital signal processing (DSP) techniques.
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Figure 5.1 Architecture of GSM Network. In order to ensure that network operators will have several sources of cellular infrastructure equipment, GSM decided to specify not only the air interface, but also the main interfaces that identify different parts. There are three dominant interfaces, namely, an interface between MSC and the base Transceiver Station (BTS), and an Um interface between the BTS and MS.
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The other key relationship in cellular networks is the one between the reuse factor K and call capacity. First of all, call capacity depends on the number of available channels. In GSM, a limited number of frequencies is available (for GSM: 124 frequencies, and for GSM-1800: 374 frequencies). The frequencies are grouped into frequency sets. If K increases, the number of frequencies per set (and so per cell) decreases, and so the call capacity per cell. The value of K in GSM cellular networks varies between 4 and 21. Note that in real networks, K is not a constant within the whole PLMN area, but varies depending on the traffic capacity needed in certain regions. Typically, K is high in urban regions and low in rural regions. If K increases, then performance increases If K increases, then call capacity decreases per cell The number of sites to cover a given area with a given high traffic density, and hence the cost of the infrastructure, is determined directly by the reuse factor and the number of traffic channels that can be extracted from the available spectrum. These two factors are compounded in what is called spectral efficiency of the system. Not all systems allow the same performance in this domain: they depend in particular on the robustness of the radio transmission scheme against interference, but also on the use of a number of technical tricks, such as reducing transmission during the silences of a speech communication. The spectral efficiency, together
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ECE with the constraints on the cell size, determines also the possible compromises between the capacity and the cost of the infrastructure. All this explains the importance given to spectral efficiency.
bandwidth with some reliability. In any cellular system or cellular technology, it is necessary to have a scheme that enables several multiple users to gain access to it and use it simultaneously. As cellular technology has progressed different multiple access schemes have been used. They form the very core of the way in which the radio technology of the cellular system works. A mix of Frequency Division Multiple Access (FDMA) and Time Division Multiple Access (TDMA), combined with frequency hopping, has been adopted as the multiple access schemes for GSM. GSM chose a combination of TDMA/FDMA as its method. The FDMA part involves the division by frequency of the total 25 MHz bandwidth into 124 carrier frequencies of 200 kHz bandwidth. One or more carrier frequencies are then assigned to each BS. Each of these carrier frequencies is then divided in time, using a TDMA scheme, into eight time slots. One time slot is used for transmission by the mobile and one for reception. They are separated in time so that the mobile unit does not receive and transmit at the same time.
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ECE amplifiers. An easy way to understand the GMSK signal is to first investigate its precursor, MinimumShift Keying (MSK).The following figure indicates the steps in generating an MSK signal. How the data is treated in GMSK is explained below: The waveforms are all aligned together in phase. Little scales are placed are placed in the figure to help make the phase relationships between the waveforms clearer. 10 bits of the data stream {1101011000} is considered for analysis. The data stream is divided into odd and even bit streams:(odd bits and even bits).In creating odd bits and even bits ,each alternate odd and even bit in data is hold for two bit times. Staggering odd bits and even bits already helps to create a waveform with minimal AM. For convenience odd bits and even bits are made to take the values 1or1. In GSM case, if the data rate (in waveform data) is 270.833 kbps, then the staggered odd bits and even bits will have half the rate135.4 kbps. The fourth and fifth wave forms in the following figure are the high freq and the low freq versions, respectively, of the carrier. Since MSK is a form of FSK, finally
modulated carrier needs two diff. Frequency components (low and high).the MSK signal is created by shifting between these two frequencies. The MSK signal is created starting with bit number 2, with the help of the truth table given below along with the waveforms. At any instant the odd and even bit values are taken from the table and follow the rules as given in the truth table to obtain the MSK waveform at that instant. Either the high or the low frequency versions of the carrier is picked corresponding to the instant under consideration and also according to the sense instructions(+or-) the wave form is to be turned up or down.
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Figure 4.4 GMSK wave forms Smooth phase transitions can be noticed, as the MSK waveform changes its frequency one from the other. These high and low frequencies shall be as close together as possible in the frequency domain. To make a GMSK signal from an MSK signal ,the stretched data waveforms (each135.4 kbps) have to be filtered with a Gaussian filter of an appropriate bandwidth defined by the BT product(Bandwidth*Time).In GSM case ,BT is 0.3,which makes B=81.3 kHz when T is 3.7 micro sec (T=1/270.833).
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ECE the time domain when it is transmitted over the radio path. Inside the receiver, the information is decompressed, or expanded, in order to regenerate the continuous audio signal. The device that transforms the human voice into a digital stream of data suitable for transmission over the radio interface and regenerates an audible analog representation of the received data (voice) is called a speech codec. 4.5.1 HOW THE SPEECH CODING WORKS IN GSM Sound (human voice) is converted to an electrical signal by the microphone. To digitize this analog signal, it is sampled at 8 KHz rate. The signal is sampled after filtering. Every 125 micro seconds, a value is sampled from the analog signal and quantized by a 13 bit word. The 125 micro sec sampling intervals are derived from a sampling frequency of 8 KHz, which are 8000 samples per second. A sampling rate of 8000 samples per second means that the output of Analog to Digital converter delivers a data rate of 8000x 13bps=104 Kbps.104 Kbps data is far too high to be economically transmitted over the radio interface; considering the Bandwidth scarcity. Band width has to be shared by number of users for costing advantages. The speech coder will have to do something to significantly reduce this rate by extracting irrelevant components in the data stream. The speech coder has to search for excess baggage we can safely remove from the bit stream scheduled for transport over the radio path. GSM uses to processes to strip redundant fat from the data representing voice traffic. The compression algorithm used in GSM is a procedure called RPE-LTP.
4.5.2 REGULAR PULSE EXCITATION AND LONG TERM PREDICTION (RPELTP) Every 20ms, 160 sampled values from the ADC are taken and stored in an intermediate memory. An analysis of a set of data samples produces eight filter coefficients and an excitation signal for a time-invariant digital filter. This filter can be regarded as a digital imitation of the human vocal tract, where the finer coefficients represent vocal modifiers(e.g., teeth, tongue, pharynx)and the excitation signal represents the sound(e.g., pitch , loudness) or the absence of sound that we pass through the vocal tract(filter). A correct setting of filter coefficients and an appropriate excitation signal yields a sound typical of the human voice. The procedure, so far, has not performed any data reductions. The reductions come in
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ECE further steps, which take advantage of certain attributes of the human ear and vocal tract .The 160 samples, transformed into filter coefficients, are divided into four blocks of 40 samples each. Each block represents a 5-ms period of voice. These blocks are sorted into four sequences. Where each sequence contains very forth sample from the original 160 samples. Sequence number 1 contains samples 1, 5, 9, 13., 37, sequence number 2 contains samples 2, 6, 10, 14, .38, Sequence number 3 contains samples3, 7, 11, 15,39, and Sequence number 4 contains samples 4, 8, 12, 1640. The first reduction in data comes when the speech encoder selects the sequence with the most energy. This linear predictive coding (LPC) and regular pulse excitation (RPE) analysis has a very short memory of approximately 1ms. A more long-term consideration of neighboring (or adjacent) blocks in time is not performed here, There are numerous correlations in the human voice, especially in long vowels such as the in car, where the same sound recurs in succeeding 5ms samples. Taking the similarity of sounds between adjacent samples (Adjacent 5-ms blocks) into account can significantly reduce the amount of data required to describe the human voice. This second reduction task is performed by a LTP Function. 4.5.3 LONG-TERM PREDICTION ANALYSIS (LTP) The LTP function accepts a sequence selected by the LPC/RPE analysis. Upon accepting sequence, it then looks among all the previous sequences passed to it (which will reside in another intermediate memory for 15ms) for the earlier sequence that has the highest correlation to ( bears the greatest resemblance to ) the current sequence. It can be said that the LTP function
looks for the one sequence from among those already received that is most similar to the sequence just received from the LPC/RPE. Now it is only necessary to transmit a value representing the difference between the two sequences, along with a pointer to tell the receiver on the other end of the radio channel, which sequence it should select among its recently received sequences for comparison. The receiver knows which differential values it has to apply to which sequences. The transmission of the whole sequence is not necessary, only the difference between sequences, This further reduces the data on the channel. The speech coder issues a block of 260bits (a speech frame) once every 20ms. This corresponds to net data rate of 13kbps, a data reduction of a factor of eight. Speech transcoding
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ECE is a task that requires a large number of calculations at high speeds. It is, therefore, an ideal application for digital signal processing (DSP) techniques.
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Figure 5.1 Architecture of GSM Network. In order to ensure that network operators will have several sources of cellular infrastructure equipment, GSM decided to specify not only the air interface, but also the main interfaces that identify different parts. There are three dominant interfaces, namely, an interface between MSC and the base Transceiver Station (BTS), and an Um interface between the BTS and MS.
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Cell site
The GSM service is the total area served by the combination of all member countries where a mobile can be serviced. The next level is the PLMN service area. There can be several within a country, based on its size. The links between a GSM/PLMN network and other PSTN, ISDN, or PLMN network will be on the level of international or national transit exchange. All incoming calls for a GSM/PLMN network will be routed to a gateway MSC. A gateway MSC works as an incoming transit exchange for the GSM/PLMN. In a GSM/PLMN network, all mobile-terminated calls will be routed to a gateway MSC. Call connections between PLMNs, or to fixed networks, must be routed through certain designated MSCs called a gateway MSC. The gateway MSC contains the interworking functions to make these connections. They also route incoming calls to the proper MSC within the network. The next level of division is the MSC/VLR service area. In one PLMN there can be several MSC/VLR service areas. MSC/VLR is a role controller of calls within its jurisdiction. In order to route a call to a mobile subscriber, the path through links to the MSC in the MSC area where the subscriber is currently located. The mobile location can be uniquely identified since the MS is registered in a VLR, which is generally associated with an MSC. The next division level is that of the LAs within a MSC/VLR combination. There are several LAs within one MSC/VLR combination. A LA is a part of the MSC/VLR service area in which a MS may move freely without updating location information to the MSC/VLR
exchange that control the LA. Within a LA a paging message is broadcast in order to find the called mobile subscriber. The LA can be identified by the system using the Location Area Identity (LAI). The LA is used by the GSM system to search for a subscriber in an active state. Lastly, a LA is divided into many cells. A cell is an identity served by one BTS. The MS distinguishes between cells using the Base Station Identification code (BSIC) that the cell site broadcast over the air.
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ECE personally assigned to one subscriber. The SIM is a subscribe module which stores all the subscriber-related information. When a subscribers SIM is inserted into the ME of an MS that MS belongs to the subscriber, and the call is delivered to that Ms. The ME is not associated with a called number-it is linked to the SIM. In this case, any ME can be used by a subscriber when the SIM is inserted in the MS. SIM is needed in order to access the services provided by the GSM PLMN. MS can be installed in Vehicles or can be portable or handheld stations. The MS may include provisions for data communication as well as voice. A mobile transmits and receives message to and from the GSM system over the air interface to establish and continue connections through the system. Different type of MSs can provide different type of data interfaces. To provide a common model for describing these different MS configuration, reference configuration for MS, similar to those defined for ISDN land stations, has been defined. Each MS is identified by an IMEI that is permanently stored in the mobile unit. Upon request, the MS sends this number over the signaling channel to the MSC. The IMEI can be used to identify mobile units that are reported stolen or operating incorrectly. Just as the IMEI identities the mobile equipment, other numbers are used to identify the mobile subscriber. Different subscriber identities are used in different phases of call setup. The Mobile Subscriber ISDN Number (MSISDN) is the number that the calling party dials in order to reach the subscriber. It is used by the land network to route calls toward an appropriate MSC. The international mobile subscribe identity (IMSI) is the primary function of the subscriber within the mobile network and is permanently assigned to him. The GSM system can also assign a Temporary Mobile Subscriber Identity (TMSI) to identity a mobile. This number can be periodically changed by the system and protect the subscriber from being identified by those attempting to monitor the radio channel. 5.3.1 FUNCTIONS OF MS The primary functions of MS are to transmit and receive voice and data over the air interface of the GSM system. MS performs the signal processing function of digitizing, encoding, error protecting, encrypting, and modulating the transmitted signals. It also performs the inverse functions on the received signals from the BS. In order to transmit voice and data signals, the mobile must be in synchronization with the
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ECE system so that the messages are the transmitted and received by the mobile at the correct instant. To achieve this, the MS automatically tunes and synchronizes to the frequency and TDMA timeslot specified by the BSC. This message is received over a dedicated timeslot several times within a multiform period of 51 frames. We shall discuss the details of this in the next chapter. The exact synchronization will also include adjusting the timing advance to compensate for varying distance of the mobile from the BTS. The MS monitors the power level and signal quality, determined by the BER for known receiver bit sequences, from both its current BTS and up to six surrounding BTSs. This data is received on the downlink broadcast control channel. The MS determines and send to the current BTS a list of the six best-received BTS signals. The measurement results from MS on downlink quality and surrounding BTS signal levels are sent to BSC and processed within the BSC. The system then uses this list for best cell handover decisions. MS keeps the GSM network informed of its location during both national and international roaming, even when it is inactive. This enables the system to page in its present LA. The MS includes an equalizer that compensates for multi-path distortion on the received signal. This reduces inter-symbol interference that would otherwise degrade the BER. Finally, the MS can store and display short received alphanumeric messages on the liquid crystal display (LCD) that is used to show call dialing and status information. These messages are limited to 160 characters in length.
Power Levels These are five different categories of mobile telephone units specified by the European
GSM system: 20W, 8W, 5W, 2W, and 0.8W. These correspond to 43-dBm, 39-dBm, and 37dBm, 33-dBm, and 29-dBm power levels. The 20-W and 8-W units (peak power) are either for vehicle-mounted or portable station use. The MS power is adjustable in 2-dB steps from its nominal value down to 20mW (13 dBm). This is done automatically under remote control from the BTS, which monitors the received power and adjusts the MS transmitter to the minimum power setting necessary for reliable transmission. 5.3.2 SIM CARD
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ECE As described in the first chapter, GSM subscribers are provided with a SIM card with its
unique identification at the very beginning of the service. By divorcing the subscriber ID from the equipment ID, the subscriber may never own the GSM mobile equipment set. The subscriber is identified in the system when he inserts the SIM card in the mobile equipment. This provides an enormous amount of flexibility to the subscribers since they can now use any GSM-specified mobile equipment. Thus with a SIM card the idea of Personalize the equipment currently in use and the respective information used by the network (location information) needs to be updated. The smart card SIM is portable between Mobile Equipment (ME) units. The user only needs to take his smart card on a trip. He can then rent a ME unit at the destination, even in another country, and insert his own SIM. Any calls he makes will be charged to his home GSM account. Also, the GSM system will be able to reach him at the ME unit he is currently using. The SIM is a removable SC, the size of a credit card, and contains an integrated circuit chip with a microprocessor, random access memory (RAM), and read only memory (ROM). It is inserted in the MS unit by the subscriber when he or she wants to use the MS to make or receive a call. As stated, a SIM also comes in a modular from that can be mounted in the subscribers equipment. When a mobile subscriber wants to use the system, he or she mounts their SIM card and provide their Personal Identification Number (PIN), which is compared with a PIN stored within the SIM. If the user enters three incorrect PIN codes, the SIM is disabled. The PIN can also be permanently bypassed by the service provider if requested by the subscriber. Disabling the PIN code simplifies the call setup but reduces the protection of the users account in the event of a stolen SIM.
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ECE A TMSI is a MSC-VLR specific alias that is designed to maintain user confidentiality. It is
assigned only after successful subscriber authentication. The correlation of a TMSI to an IMSI only occurs during a mobile subscribers initial transaction with an MSC (for example, location updating). Under certain condition (such as traffic system disruption and malfunctioning of the system), the MSC can direct individual TMSIs to provide the MSC with their IMSI. 5.4.3 MOBILE STATION ISDN NUMBER (MSISDN) The MS international number must be dialed after the international prefix in order to obtain a mobile subscriber in another country. The MSISDN numbers is composed of the country code (CC) followed by the National Significant Number (NSN), which shall not exceed 15 digits. 5.4.4 MOBILE STATION ROAMING NUMBER (MSRN) The MSRN is allocated on temporary basis when the MS roams into another numbering area. The MSRN number is used by the HLR for rerouting calls to the MS. It is assigned upon demand by the HLR on a per-call basis. The MSRN for PSTN/ISDN routing shall have the same structure as international ISDN numbers in the area in which the MSRN is allocated. The HLR knows in what MSC/VLR service area the subscriber is located. At the reception of the MSRN, HLR sends it to the GMSC, which can now route the call to the MSC/VLR exchange where the called subscriber is currently registered.
5.4.5 INTERNATIONAL MOBILE EQUIPMENT IDENTITY (IMEI) The IMEI is the unique identity of the equipment used by a subscriber by each PLMN and is used to determine authorized (white), unauthorized (black), and malfunctioning (gray) GSM hardware. In conjunction with the IMSI, it is used to ensure that only authorized users are granted access to the system. An IMEI is never sent in cipher mode by MS.
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ECE one or more cells. A BSS may consist of one or more BTS. The interface between BSC and BTS is designed as an A-bis interface. The BSS includes two types of machines: the BTS in contact with the MSs through the radio interface and the BSC, the latter being in contact with the MSC. The function split is basically between transmission equipment, the BTS, and managing equipment at the BSC. A BTS compares radio transmission and reception devices, up to and including the antennas, and also all the signal processing specific to the radio interface. A single transceiver within BTS supports eight basic radio channels of the same TDM frame. A BSC is a network component in the PLMN that function for control of one or more BTS. It is a functional entity that handles common control functions within a BTS. A BTS is a network component that serves one cell and is controlled by a BSC. BTS is typically able to handle three to five radio carries, carrying between 24 and 40 simultaneous communication. Reducing the BTS volume is important to keeping down the cost of the cell sites. An important component of the BSS that is considered in the GSM architecture as a part of the BTS is the Transcoder/Rate Adapter Unit (TRAU). The TRAU is the equipment in which coding and decoding is carried out as well as rate adoption in case of data. Although the specifications consider the TRAU as a subpart of the BTS, it can be sited away from the BTS (at MSC), and even between the BSC and the MSC. The interface between the MSC and the BSS is a standardized SS7 interface (Ainterface) that, as stated before, is fully defined in the GSM recommendations. This allows the
system operator to purchase switching equipment from one supplier and radio equipment and the controller from another. The interface between the BSC and a remote BTS likewise is a standard the A-bis. In splitting the BSS functions between BTS and BSC, the main principle was that only such functions that had to reside close to the radio transmitters/receivers should be placed in BTS. This will also help reduce the complexity of the BTS. 5.5.1 FUNCTIONS OF BTS
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As stated, the primary responsibility of the BTS is to transmit and receive radio signals from a mobile unit over an air interface. To perform this function completely, the signals are encoded, encrypted, multiplexed, modulated, and then fed to the antenna system at the cell site. Trans-coding to bring 13-kbps speech to a standard data rate of 16 kbps and then combining four of these signals to 64 kbps is essentially a part of BTS, though it can be done at BSC or at MSC. The voice communication can be either at a full or half rate over logical speech channel. In order to keep the mobile synchronized, BTS transmits frequency and time synchronization signals over frequency correction channel (FCCH and BCCH logical channels. The received signal from the mobile is decoded, decrypted, and equalized for channel impairments. Random access detection is made by BTS, which then sends the message to BSC. The channel subsequent assignment is made by BSC. Timing advance is determined by BTS. BTS signals the mobile for proper timing adjustment. Uplink radio channel measurement corresponding to the downlink measurements made by MS has to be made by BTS.
5.5.2 BTS-BSC CONFIGURATIONS There are several BTS-BSC configurations: single site; single cell, single site, multi cell and multisite, multi cell. These configurations are chosen based on the rural or urban application. These configurations make the GSM system economical since the operation has options to adapt the best layout based on the traffic requirement. Thus, in some sense, system optimization is possible by the proper choice of the configuration. These include Omni directional rural
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ECE configuration where the BSC and BTS are on the same site; chain and multi drop loop configuration in which several BTSs are controlled by a single remote BSC with a chain or ring connection topology; rural star configuration in which several BTSs are connected by individual lines to the same BSC and sectored urban configuration in which three BTSs share the same site and are controlled by either a collocated or remote BSC. In rural areas, most BSCs are installed to provide maximum coverage rather than maximum capacity.
5.6.1 BASE STATION CONTROLLER (BSC) The BSC is connected to the MSC on one side and to the BTS on the other. The BSC performs the Radio Resource (RR) management for the cells under its control. It assigns and release frequencies and timeslots for all MSs in its own area. The BSC performs the inter cell handover for MSs moving between BTS in its control. It also reallocates frequencies to the BTSs in its area to meet locally heavy demands during peak hours or on special events. The BSC
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ECE controls the power transmission of both BSSs and MSs in its area. The minimum power level for a mobile unit is broadcast over the BCCH. The BSC provides the time and frequency synchronization reference signals broadcast by its BTSs. The BSC also measures the time delay of received MS signals relative to the BTS clock. If the received MS signal is not centered in its assigned timeslot at the BTS, The BSC can direct the BTS to notify the MS to advance the timing such that proper synchronization takes place. The functions of BSC are as follows. The BSC may also perform traffic concentration to reduce the number of transmission lines from the BSC to its BTSs, as discussed in the last section.
and identities related to the same MS and contained in different registers: IMSI, TMSI, ISDN number, and MSRN. In general identities are used in the interface between the MSC and the MS, while numbers are used in the fixed part of the network, such as, for routing. 5.7.2 FUNCTIONS OF MSC As stated, the main function of the MSC is to coordinate the setup of calls between GSM mobile and PSTN users. Specifically, it performs functions such as paging, resource allocation,
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location registration, and encryption. Specifically, the call-handling function of paging is controlled by MSC. MSC coordinates the setup of call to and from all GSM subscribers operating in its areas. The dynamics allocation of access resources is done in coordination with the BSS. More specifically, the MSC decides when and which types of channels should be assigned to which MS. The channel identity and related radio parameters are the responsibility of the BSS; The MSC provides the control of interworking with different networks. It is transparent for the subscriber authentication procedure. The MSC supervises the connection transfer between different BSSs for MSs, with an active call, moving from one call to another. This is ensured if the two BSSs are connected to the same MSC but also when they are not. In this latter case the procedure is more complex, since more than one MSC in involved. The MSC performs billing on calls for all subscribers based in its areas. When the subscriber is roaming elsewhere, the MSC obtains data for the call billing from the visited MSC. Encryption parameters transfers from VLR to BSS to facilitate ciphering on the radio interface are done by MSC. The exchange of signaling information on the various interface toward the other network elements and the management of the interface themselves are all controlled by the MSC. Finally, the MSC serves as a SMS gateway to forward SMS messages from Short Message Service Centers (SMSC) to the subscribers and from the subscribers to the SMSCs. It thus acts as a message mailbox and delivery system. 5.7.3 VISITOR LOCATION REGISTER (VLR) The VLR is collocated with an MSC. A MS roaming in an MSC area is controlled by the VLR responsible for that area. When a MS appears in a LA, it starts a registration procedure. The MSC for that area notices this registration and transfers to the VLR the identity of the LA where the MS is situated. A VLR may be in charge of one or several MSC LAs. The VLR constitutes the databases that support the MSC in the storage and retrieval of the data of subscribers present in its area. When an MS enters the MSC area borders, it signals its arrival to the MSC that stores its identify in the VLR. The information necessary to manage the MS is contained in the HLR and is transferred to the VLR so that they can be easily retrieved if so required. Data Stored in VLR The data contained in the VLR and in the HLR are more or less the same. Nevertheless
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ECE the data are present in the VLR only as long as the MS is registered in the area related to that VLR. Data associated with the movement of mobile are IMSI, MSISDN, MSRN, and TMSI. The terms permanent and temporary, in this case, are meaningful only during that time interval. Some data are mandatory, others are optional. 5.7.4 HOME LOCATION REGISTER (HLR) The HLR is a database that permanently stores data related to a given set of subscribers. The HLR is the reference database for subscriber parameters. Various identification numbers and addresses as well as authentication parameters, services subscribed, and special routing information are stored. Current subscriber status including a subscribers temporary roaming number and associated VLR if the mobile is roaming, are maintained. The HLR provides data needed to route calls to all MS-SIMs homes based in its MSC area, even when they are roaming out of area or in other GSM networks. The HLR provides the current location data needed to support searching for and paging the MS-SIM for incoming calls, wherever the MS-SIM may be. The HLR is responsible for storage and provision of SIM authentication and encryption parameters needed by the MSC where the MS-SIM is operating. It obtains these parameters from the AUC. The HLR maintains record of which supplementary service each user has subscribed to and provides permission control in granting services. The HLR stores the identification of SMS gateways that have messages for the subscriber under the SMS until they can be transmitted to the subscriber and receipt is knowledge. Some data are mandatory, other data are optional. Both the HLR and the VLR can be implemented in the same equipment in an MSC (collocated). A PLMN may contain one or several HLRs. 5.7.5 AUTHENTICATION CENTER (AUC) The AUC stores information that is necessary to protect communication through the air interface against intrusions, to which the mobile is vulnerable. The legitimacy of the subscriber is established through authentication and ciphering, which protects the user information against unwanted disclosure. Authentication information and ciphering keys are stored in a database within the AUC, which protects the user information against unwanted disclosure and access. In the authentication procedure, the key Ki is never transmitted to the mobile over the air
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ECE path, only a random number is sent. In order to gain access to the system, the mobile must provide the correct Signed Response (SRES) in answer to a random number (RAND) generated by AUC. Also, Ki and the cipher key Kc are never transmitted across the air interface between the BTS and the MS. Only the random challenge and the calculated response are transmitted. Thus, the value of Ki and Kc are kept secure. The cipher key, on the other hand, is transmitted on the SS7 link between the home HLR/AUC and the visited MSC, which is a point of potential vulnerability. On the other hand, the random number and cipher key is supposed to change with each phone call, so finding them on one call will not benefit using them on the next call. The HLR is also responsible for the authentication of the subscriber each time he makes or receives a call. The AUC, which actually performs this function, is a separate GSM entity that will often be physically included with the HLR. Being separate, it will use separate processing equipment for the AUC database functions.
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IWF to provide this interfacing capability. The IWF, which in essence is a part of MSC, provides the subscriber with access to data rate and protocol conversion facilities so that data can be transmitted between GSM Data Terminal Equipment (DTE) and a land-line DTE.
Echo Canceller (EC)
EC is used on the PSTN side of the MSC for all voice circuits. The EC is required at the MSC PSTN interface to reduce the effect of GSM delay when the mobile is connected to the PSTN circuit. The total round-trip delay introduced by the GSM system, which is the result of speech encoding, decoding and signal processing, is of the order of 180 ms. Normally this delay would not be an annoying factor to the mobile, except when communicating to PSTN as it requires a two-wire to four-wire hybrid transformer in the circuit. This hybrid is required at the local switching office because the standard local loop is a two-wire circuit. Due to the presence of this hybrid, some of the energy at its four-wire receive side from the mobile is coupled to the four-wire transmit side and thus retransmitted to the mobile. This causes the echo, which does not affect the land subscriber but is an annoying factor to the mobile. The standard EC cancels about 70ms of delay. During a normal PSTN (land-to-land call), no echo is apparent because the delay is too short and the land user is unable to distinguish between the echo and the normal telephone side tones However, with the GSM round-trip delay added and without the EC, the effect would be irritating to the MS subscriber.
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Figure 5.3 Operation and Maintenance Centre. The fault management functions of the OSS allow network devices to be manually or automatically removed from or restored to service. The status of network devices can be checked, and tests and diagnostics on various devices can be invoked. For example, diagnostics may be initiated remotely by the OSS. A mobile call trace facility can also be invoked. The performance management functions included collecting traffic statistics from the GSM network entities and archiving them in disk files or displaying them for analysis. Because a potential to collect large amounts of data exists, maintenance personal can select which of the detailed statistics to be collected based on personal interests and past experience. As a result of performance analysis, if necessary, an alarm can be set remotely. The OSS provides system change control for the software revisions and configuration data bases in the network entities or uploaded to the OSS. The OSS also keeps track of the different software versions running on different subsystem of the GSM.
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ECE Mobility management entails the GSM system keeping track of the mobile while on the move.
The mobility management is implemented through mobility management (MM) sub-layer which is present in layer 3 of the protocol stack at MS and MSC. The functions performed by the mobility management are: Subscribe Data management at MSC/VLR-
Subscriber data from HLR are retrieved by MM at the time of first location updating of the subscriber. Dynamic data change for a subscribe is also managed by the MM. Services provided to upper layersMM provides basic means of transportation of upper CM sub-layer messages between MS and the network Handover procedures ensure smooth transition from one radio network to another. Subscriber Authentication and confidentiality Management MM procedures ensure data confidentiality of a subscriber MM procedures provide a means for to ensure data confidentiality at radio interface. Mobility management is implemented through MM procedures, which are broadly classified in to two groups I) MM Common Procedure II) MM Specific Procedure A MM specific procedure can only be started if no other MM specific procedure is running or no MM connection exists between the network and the mobile station. The end of the running MM connection has to be awaited before a new MM specific procedure can be started. During the lifetime of a MM specific procedure, if a MM connection establishment is requested by a CM entity this request will either be rejected or be delayed until the running MM
specific procedure is terminated (this depends upon implementation). Any MM common procedure (except IMSI detach) may be started during MM specific procedure.
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ECE The purpose of the TMSI reallocation procedures is to provide identify confidentiality i.e. to
protect a user against being identified and located by an intruder. TMSI is used for identification within the radio interface signaling procedures instead of IMSI. Usually the TMSI reallocation is performed at least at each change of a location area. The reallocation of TMSI can be performed explicitly after predetermined no. of accesses by MS to the network or implicitly by a location updating procedure. TMSI reallocation can be initiated by the network at any time whilst RR connection exists between the network and the mobile station. In case of TMSI reallocation procedure initiated by the network, network sends TMSI reallocation command message to the MS containing new TMSI/LAI MS on receiving the message stores new TMSI and LAI in SIM and deletes the older entries and sends TMSI reallocation complete message to the network. 2. AUTHENTICATION PROCEDURE Authentication Triplets: At network side, authentication procedure requires authentication triplets. Authentication triplet consists of: Random number RAND (128 bits) Signed response SRES (32 bits) Ciphering key (64 bits)
While initiating authentication procedure, if network has no authentication triplet or all triplets have been used, it requests AUC for the same. The index of currently used triplet is known as CKSN. (Ciphering key sequence number). 3. IDENTIFICATION PROCEDURE The identification procedure is used by the network to request a MS to provide specific identification parameters to the network e.g. IMSI, IMEI.
In case MS update location in the system using TMSI, but due to data base failure, TMSI at network end is no more available; network initiates identification procedure and asks for IMSI. If network is unable to receive identity response, it clears all the ongoing MM connections and releases radio resources.
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4. IMSI DETACH PROCEDURE The purpose of this procedure is to indicate the network that MS has switched off. This enables the network not to page for the subscriber and invokes other applicable supplementary service (e.g. call forwarding etc.) 5. CIPHERING PROCEDURE Ciphering procedure is needed to encrypt data transmission over radio interface. When MSC needs to indicate ciphering on radio interface, it sends Cipher Mode Command message to BSS. This message contains Kc and a list of permitted algorithms. BSS stores Kc for this session and sends Cipher mode message to the MS. BSS enable encryption and uses Kc and permitted algorithm to encrypt /decrypt data. In case ciphering is completed successfully the network receives a Cipher Mode Complete command from the MS. This message contains the algorithm used for ciphering. MSC stores this information and proceeds for further activities like call set- up etc. In case MS is not able to support the ciphering it sends Cipher Mode Reject command to the network. At MSC the encryption control is operator controlled. If ciphering is mandatory and network receives a Cipher Mode Reject command from the MS, MSC clears all the ongoing MM connection and then releases radio resources. 6. ABORT PROCEDURE The procedure is used to abort any ongoing or established MM connection.
Periodic location updating -- It is used to periodically notify the availability of the MS to the network.
IMSI attach procedure It is used to indicate the IMSI as active in the network.
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to the network. MS starts a guard timer and enters state location updating initiated. After this, the network initiates authentication and ciphering procedure. After successful authentication and ciphering location updating procedure proceeds further. To limit the number of unsuccessful location updating attempts, an Attempt Counter is maintained at MS. Attempt counter is incremented when a location updating procedure fails. Attempt counter is reset when MS is powered on /a SIM is inserted / location update is successfully completed. If the Location updating is successfully accepted by the network a Location Updating Accept message is transferred to the MS. Implicit TMSI reallocation procedure is also invoked. MS on receiving the Location Updating Accept stores the received LAI, stops the guard timer, reset the attempt counter and sets the update status of SIM to update. MS at all times maintains a list of forbidden LAIs and PLMNs in SIM. If the LAI or PLMN identity contained in the Location Updating Accept message is a member of any of the forbidden lists then any such entries are deleted in MS. After location-updating procedure is over, the RR connection is released. The network initiates the release. MS waits for the release and if within time out the connection is not released, MS aborts it. MM procedures make use of certain MAP (Mobile Application Part). For example location updating procedure, which is an MM specific procedure make use of MAP procedures like: Down-loading of subscriber related data from HLR to VLR through MAP procedure on C interface. Thus LU procedure of MM makes use of a MAP procedure also. Similarly authentication procedure, which is a MM common procedure, makes use of MAP procedure to retrieve authentication triplets from AuC.
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ECE service to MM sub-layer could be-call control (CC), short message service (SMS) or
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supplementary service (SS). An MM connection is established and released on request from CM entities. Different CM entity communicates with their pier entities using different MM
connection. An MM connection requires an RR connection. Several MM connections may be active at the same time and all simultaneous MM connection for a given MS use the same RR connection. Different MM connections are identified by different protocol discriminator (PD) and transaction identifier (TI) value. MM connection establishment may be initiated by the MS or the network. For an MS to initiate MM connection its updating status should be updated and MM sublayer should be either in IDLE or ACTIVE state. If any MM specific procedure is running then a new MM connection establishment will either be rejected or delayed (depending upon implementation). If no RR connection exists between MS and network, RR connection is established by sending CM service request message to the network and MM sub- layer enters wait for o/g MM Connection State. If RR connection already exists and an MM connection is active, CM service Request message is sent and MM sub-layer enters wait for additional o/g MM connection state. The CM service Request message contains the type of CM service requested (o/g call, Emergency call, SMS, SS). If the network can accept the CM service request, a CM Service Accept message is sent to MS and on recovering this message MM connection become active and CC message can be transferred by CM entity. If network cannot accept CM service request from MS, a CM service Reject message is sent to MS. In case of MT (mobile terminated) call (incoming voice call or short message to the MS) CM sublayer entity in the network request MM sub layer to establish a MM connection. The MM sub-layer is informed after completion of paging procedure. Now these can be two scenarios depending upon whether RR connection to the desired MS already exists or not.
(a) RR connection to the desired MS already exists This could be the case when MS is in conversation mode and there is a short message for the MS. When RR connection between the network and desired MS exists and also no MM specific
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ECE procedure is running (is no location updating etc. is in progress) the network establishes new MM
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connection over same RR connection with new PD/TI combination. Before establishment of a new MM connection the network may initiate any of MM common procedure like authentication, ciphering etc. and wait for their successful completion. (b) RR connection to the desired MS does not exist MM sub layer first requests to establish an RR connection and on connection establishment MM sub-layer may initiate any of MM common procedure (authentication, ciphering etc.) Upon successful completion of any such procedure MM sub layer informs the requesting CM sub-layer entity. If RR connection establishment is unsuccessful or any of the MM common procedure fails, this is indicated to CM sub-layer with appropriate error cause. A CM sub-layer entity, after having been advised that a MM connection has been established, requests the transfer of CM massages. The CM messages passed to MM sub-layer are sent to these other side of the interface with PD & TI set according to source entity. Upon receiving CM message, the MM sub-layer on the side of the interface distributes it to the relevant CM entity according to the PD & TI value. After the information transfer between CM entities is over, an established MM connection can be released by local CM entity. The release of CM connection is then done locally in the MM sub-layer. After the release of last MM connection by its user, the MM sub-layer decides to release RR connection requesting RR sub-layer.
6.5.1 RF CHANNEL OVERVIEW RF channel play important role in call processing case. These are basically three types of RF control channel.
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Broadcast control channel The broadcast channels are points to multi-point channel, which are defined only for down-link direction (BTS to mobile station). They are divided into: BCCH (Broad cast control channel): BCCH acts as a beacon. It informs the mobile about system configuration parameters . Using this information MS choose the best cell to attach to. BCCH is always transmitted on full power and it is never frequency hopped. FCCHC (Frequency correction channel): MS must tune to FCCH to listen to BCCH. FCCH transmits a constant frequency shift of the radio carrier that is used by the MS for frequency correction. SCH (synchronization channel): SCH is used to synchronize the MS in time .SCH carries TDMA frame number and BSIC (Base Station Identity Code)
Common control channels Common control channels are specified as point to multi-point, which operate only in one direction either in up-link or down-link direction. PCH (Paging Channel): PCH is used in down-link direction for sending paging message to MS whenever there is incoming call. RACH (Random Access Channel): RACH is used by the MS to request allocation of a specific dedicated control channel (SDCCH) either in response to a paging message or for call origination /registration from the MS. This is an up-link channel and operates in point to point mode. AGCH (Access Grant Channel): AGCH is a logical control channel which is used to allocate a specific dedicated control channel (SDCCH) to MS when MS request for a channel over RACH. AGCH is used in downlink direction.
Dedicated Control Channel Dedicated control channel are full duplex, point to point channel. They are used for signaling between the BTS and certain MS. They are divided into
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ECE SACCH (Slow Associated Control Channel): the SACCH is a duplex channel, which is always allocated to TCH or SDCCH. The SACCH is used for Radio link supervision measurements. Power control. Timing advance information. used
In 26 frame traffic multi-frame 13th frame (frame no .12) is used for SACCH.SACCH is only for non-urgent procedures.
FACCH (Fast Associated Control Channel): FACCH is requested in case the requirement of signaling is urgent and signaling requirement cannot be met by SACCH. This is the case when hand-over is required during conversation phase. During the call FACCH data is transmitted over allocated TCH instead of traffic data. This is marked by a flag known as stealing flag.
SDCCH (Stand Alone Dedicated Control Channel): The SDCCH is a duplex, point to point channel which is used for signaling in higher layer. It carries all the signaling between BTS & MS when no TCH is allocated to MS. The SDCCH is used for service request, location updates, subscriber authentication, ciphering. Equipment validation and assignment of a TCH.
6.5.2 MOBILE ORIGINATED (MO) CALL There are four distinct phase of a mobile originated call Setup phase. Ringing phase. Conversation phase. Release phase.
Out of these phases the setup phase is the most important phase and includes authentication of the subscriber, Ciphering of data over radio interface, validation of mobile equipment, validation of subscriber data at VLR for requests service and assignment of a voice channel on A-interface by MSC. Whenever MS wants to initiate on outgoing call or want to send an SMS it requested for
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dedicated control channel (SDCCH) to MS over access grant channel (AGCH). Once a SDCCH has been allocated to MS all the call set up information flow takes place over SDCCH. A connection management (CM) entity initiates a CM Service Request message to the network. Network tries to establish MM connections between the MS and the network and upon successful establishment of MM connection a CM Service Accept message is received by MS from the network. MS now sends a Call Set up Request to the network which contains the dialed digits (DD) of the called party. As the call setup message is received at the MSC/VLR certain check are performed at MSC/VLR like- whether the requested service is provisioned for the subscriber or not, whether the dialed digits are sufficient or not, any operator determined barring (ODB) does not allow call to proceed further etc. As these checks are performed at MSC/VLR a Call Proceeding Message is sent from the network towards the MS. After all the checks are successfully passed MSC sends Assignment command to the BSS which contains a free voice channel on A-interface On getting this message BSS allocates a free TCH to the MS and informs the MS to attach to it. MS on attaching to this TCH informs the BSS about it. On receiving a response from the BSS, MSC switches the speech path toward the calling MS. Thus at the end of Assignment the speech path is through from MS to MSC. It is important to note that at this stage mobile has not connected user connection as yet. MS at this stage does not listen anything. After assignment MSC sends a network set-up message to the PSTN requesting that a call be set up. Included in the message are the MS dialed digits (DD) and details specifying which trunk should be used for the call. The PSTN may involve several switching exchanges before finally reaching the final local exchange responsible for applying the ringing tone to the destination phone. The local exchange will generate the ringing tone over the trunk, or series of trunk (if several intermediate switching exchange are involved), to the MSC. At this point in time MS will hear ringing tone. The PSTN notifies the MSC with a network-alerting message when this event occurs. MSC informs the MS that the destination number is being alerted. It is
important to note that this is primarily a status message to the MS. The MS hears the ringing tone from the destination local exchange through the established voice path.
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ECE When the destination party goes off hook, PSTN informs the MSC of this event. At this
point, MS is connected to the destination party and billing is started. MSC informs the MS that connection has been established and MS acknowledges the receipts of the connect message. Under normal condition, the termination of a call is MS initiated or network initiated. In this scenario, we have assumed that MS initiates the release of the call by pressing end button and MS send a disconnect message to the MSC. The PSTN party is notified of the termination of the call by a release message from the MSC. The end- to- end connection is terminated. When MSC is left with no side task (e.g. charging indication etc.) to complete a release message is sent to the MS. MS acknowledges with a release complete message. All the resources between MSC and the MS are released completely. 6.5.3 MOBILE TERMINATED (MT) CALL The different phases of a mobile terminated call are Routing analysis. Paging. Call setup. Call release.
The phases of mobile terminated (MT) call are similar to a mobile originated (MO) call except routing analysis and paging phase. Call to a mobile subscriber in a PLMN first comes to gateway MSC (GMSC). GMSC is the MSC, which is the capable of querying HLR for subscriber routing information. GMSC need not to be part of home PLMN, though it is normal practice to have GMSC as part of PLMN in commercially deployed networks. GMSC opens a MAP (Mobile Application Part) dialogue towards HLR and Send / Routing / Info-Request (SRI request) specific service message is sent to HLR. SRI request contains MSISDN of the subscriber. HLR based on location information of this subscriber in its database, opens a MAP dialogue towards VLR and sends Provide / Roaming / Number-request (PRN request)to the VLR. VLR responds to PRN request with PRN response message, which carries an
MSRN(mobile subscriber roaming number), which can be used for routing toward visiting MSC in the network. HLR returns MSRN to GMSC (MSC that queried HLR) in SRI response message.
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ECE On getting MSRN the GMSC routes the call towards VMSC the purpose of this entire exercise is
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to locate where the terminating mobile subscriber is. The MSRN received at GMSC is in international format (Country Code + Area Code + subscriber number). Normally, based on the routing info at GMSC, the call may be routed out of the GMSC towards VMSC of the terminating subscriber, in which case appropriate signalling protocol (MF or ISUP) depending on the nature of connecting of GMSC with subsequent exchange along the route will apply. If at VMSC the terminating mobile subscriber is found to be free (idle), paging is initiated for terminating mobile subscriber. MSC uses the LAI provided by the VLR to determine which BSSs should page the MS. MSC transmit a message to each of these BSS requesting that a page be performed. Included in the message is the TMSI of the MS. Each of the BSSs broadcasts the TMSI of the mobile in a page message on paging channel (PCH). When MS detects its TMSI broadcast on the paging channel, it responds with a channel request message over Random Access Channel (RACH). Once BSS receives a channel request message, it allocates a stand alone Dedicated Control Channel (SDCCH) and forwards this channel assignment information to the MS over Access Grant Channel (AGCH). It is over this SDCCH that the MS communicates with the BSS and MSC until a traffic channel assigned to the MS. MS transmits paging response message to the BSS over the SDCCH. Included in this message is MS TMSI and LAI. BSS forwards this paging response message to the MSC. Now Authentication and Ciphering phases are performed to check the authenticity of MS and encrypt data over radio interface. On the network side after paging is initiated, while waiting for paging response, a defensive timer called, Early ACM timer is run at MSC to avoid network timeouts. On successfully getting paging response, a setup message is constructed to be sent towards terminating MS. In case paging fails due to authentication failure or when the subscriber is out of radio-coverage, the call is cleared. In case CLIP is not subscribed by the terminating mobile subscriber, calling number is not included in set-up message. In case CLIP is subscribed and PI value in calling number parameter indicates presentation allowed the number is included in the set-up message. In case CLIP is
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number is included only if CLIRO is also subscribed to. MS on receiving the set-up message performs compatibility checking before responding to the set-up message it is possible that MS might be incompatible for certain types of call setups. Assuming that MS passes compatibility checking, it acknowledges the call setup with set-up confirm message. After getting set-up confirm message from the MS, MSC performs assignment phase (similar to one discussed in MO call) and a voice path is established from MSC to the MS. MS begins altering the user after it receives the traffic channel assignment. MS send alerting message to the MSC .MSC upon receiving the alerting indication from the MS begins generating an audible ringing tone to the calling party and sends a network alerting via GMSC to the PSTN. Prior to this the calling party heard silence. At this point in the call, MS is alerting the called party by generating on audible tone. One of the three events can occur-calling party hangs-up, mobile subscriber answers the phone, or the MSC times out waiting for the mobile subscriber to the answer the call. Since radio traffic channel is a valuable resource, GSM does not allow a MS to ring forever. In the present scenario we have assumed that the mobile subscriber answers the phone. The MS in response to this action stops alerting and sends a connect message to the MSC. MSC removes the audible tone to the PSTN and connects the PSTN trunk to BSS trunk (terrestrial channel) and sends a connect message via GMSC to the PSTN. The caller and the called party now have a complete talk path. This event typically marks the beginning of the call for billing purposes. MSC sends connect acknowledge message to the MS. The release triggered by the land user is done in similar way as the release triggered by mobile user. MSC receives a release message from the network to terminate end-to-end connection. PSTN stops billing the calling landline subscriber. MSC sends a disconnect message towards the MS and MS responds by a Release message. MSC release the connection to the PSTN and acknowledges by sending a Release Complete message to PSTN. Now the voice trunk between MSC and BSS is cleared, traffic channel (TCH) is released and the resources are completely released.The mobile-to-mobile call scenario is a combination of phases encountered in mobile originated (MO) and mobile terminated (MT) call.
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CHAPTER 7 GPRS
7.1 INTRODUCTION TO GPRS
In response to customer demand for wireless Internet access and as a stepping-stone to 3G networks many GSM operators are rolling out general packet radio service (GPRS). This technology increases the data rates of existing GSM networks, allowing transport of packet based data. New GPRS handsets will be able to transfer data at rates much higher than the 9.6 or 14.4 kbps currently available to mobile-phone users. Under ideal circumstances, GPRS could support rates to 171.2 kbps, surpassing ISDN access rates. However, a more realistic data rate for early network deployments is probably around 40 kbps using one uplink and three downlink timeslots. Unlike circuit-switched 2G technology, GPRS is an always-on service. It will allow GSM operators to provide high speed Internet access at a reasonable cost by billing mobile-phone users for the amount of data they transfer rather than for the length of time they are connected to the network. This application note looks in detail at the challenges of measuring and optimizing GPRS networks. It builds directly on the first paper in this series, 'Understanding General Packet Radio Service (GPRS), AN-1377, which explains the new protocols, procedures, and other technology changes that GPRS introduces to GSM networks.
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measuring the parameters directly perceived by the user (such as throughput and delay). GPRS layer measurements are made at the layers below the application layer (for example, at the RLC/MAC/LLC layers) and are hidden to the user. These measurements offer insight about events on the GPRS layers that can impact the application layer performance. In both cases, the measurements are made using a test mobile connected to a laptop PC with special data measurement software. We also need a server for end-to-end data measurements. Signal Quality This category consists of physical layer measurements and a subset of RLC and MAC layer measurements. The measurements are made using a test mobile phone. RF Performance This category consists of network-independent measurements such as interference, scanning, and spectrum analysis. The measurements require sophisticated RF test tools such as DSP-based RF measuring receivers.
Measurement model
Data performance
Signal quality
RF performance
Application layer
GPRS layers
Phone-based
Receiver-based
Figure 7.1. GPRS drive test measurement model The relationship between the GPRS measurement model for data performance and the various protocol stacks is shown in Figure 7.2. Performance is measured at three layers: end-to-
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Server
Figure 7.2. GPRS measurement model on the protocol stacks End-to-end data performance Data performance at the application layer is measured end-to-end; that is, we simulate a real world data pattern and send it to the other end a test server, which performs the measurements and stores or sends the results back to the client, a mobile phone (MS). These measurements are made to quantify the user perception of data performance, and they are analogous to the voice quality measurements in GSM networks. We can also get information on the IP layer, depending on the type of server used and where it is placed in the network. GPRS layer data performance The data from the application layers is first processed by the GPRS layers and headers are added before it is sent onto the air interface. To a certain extent (depending on quality of service levels), the GPRS layers are capable of providing data performance information (such as LLC and RLC layer performance).
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Signal quality and RF performance These measurements are primarily physical layer measurements that provide signal level and quality information. The category may include optimization measurements such as interference monitoring and scanning. Using drive test tools, we can make these measurements simultaneously. Consequently, when low application throughput is measured, the GPRS layers, signal quality, and RF performance measurements help to determine the cause of the problem. 7.3 DATA PERFORMANCE MEASUREMENTS Classes of service Customers have different data usage requirements. Some need only low data rates for specific Web-based transactions; others need moderate data rates for applications such as email; and still others need high data rates for tasks such as transferring or downloading large, Web-based files. Often customers have different requirements at different times. GPRS has the flexibility to support dynamic management of network resources and therefore different service levels. Customers will have the option of deciding what quality of service they need and will know (at least in theory) what they are paying for. Service level choices will likely be divided into a hierarchy of quality bands or classes such as platinum, gold, silver, and bronze. On the system side, ETSI has defined a set of quality of service (QoS) classes to support the implementation of tiered service levels: Precedence class Reliability class delay class Peak throughput class mean throughput class These QoS parameters will play a key role in ensuring that customers get the data performance they expect.
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Measurement objectives Data performance is measured at the application and GPRS layers. Each set of measurements has its own objectives. At the application layer A goal of performance measurements at this layer is to simulate the user model, first by simulating the data applications (such as Web browsing, e-mail, and file transfers), and then by modeling the data load. We need to simulate the asymmetrical data transfer pattern (more data in the downlink, less in the uplink) of real world applications. A second objective is to provide real time stamped measurements. This requires getting performance information in real time. In order to quantify and benchmark these measurements, we need to refer to standards such as ETSI's QoS parameters. At present these parameters are focused primarily on the application layer, and so benchmarking GPRS network performance against these standards will provide a good way to quantify user perception of data services. At the GPRS layers Performance measurements at the GPRS layers are more complex, because many protocol layers are working simultaneously (for example, the RLC, MAC, and LLC). To identify a protocol layer of concern, we need to simulate the different QoS levels and carry out measurements at the different layers. Because data packets travel through different nodes in the GPRS network (BSS, SGSN, GGSN, PDN), we use GPRS layer measurements to identify the problem nodes. One additional objective of GPRS layer measurements is to provide control over the protocol layers. Application layer measurements We need to understand the data performance model at the application layer of testing before we delve into the bits and pieces of the performance parameters. If we take a broad view of the GPRS data communication model, we see the application layer sitting on top. An application at one end (or node) of the network the GGSN, for example communicates with a similar application at the other end, perhaps using TCP or UDP
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protocols for acknowledged or unacknowledged modes of transfer, respectively. In this model, the application layer at the mobile station (MS) communicates with the IP layer of the MS and passes the application-layer datagram to the IP layer. This IP layer (which is standard TCP/IP) forwards the information to the GGSN by way of the different GPRS nodes and protocol layers. At the GGSN, the datagram information received from the MS is returned to the IP datagram level. Then the IP layer at the GGSN communicates with the IP layer at the other end of the call (at the PC) through the Public Data Network (PDN) IP interfaces. When we measure data performance at the application layer, we want to send data from one end of our network and measure its performance at the other end, with the goal of understanding the end-to-end performance as experienced by the user.
Figure7.3. Data transfer at the Application and IP layers. Measurement configuration at the application layer There are two ways to carry out data performance measurements at the application layer: the loopback method and the end-to-end measurement method.
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Loopback measurement This approach requires a fixed unit at the network node, which collects the data coming in from the mobile and then sends the data back to the mobile (creating a loopback). The mobile compares the data sent with the data received. The process also can be reversed by simulating data traffic and sending it from the fixed unit at the network to the mobile and then back to the network. This approach raises several issues. For example, in drive testing the network, the objective is to make real time measurements, and we are interested in position-specific performance data. The loopback approach does not precisely support real world application models such as simultaneous, asymmetrical data transfer. Rather than measure uplink and downlink separately, it measures uplink and downlink as a total loop. Consequently, we do not know whether a problem exists on the uplink or the downlink. End-to-end measurements At the application layer, end-to-end measurements can be described as follows: One node transmits data and another node receives the data and measures its performance. For our test purposes, one end node is the mobile and the other is a measurement server. This server can be a located at the GGSN or somewhere in the PDN (Internet world). Since the measurements are made end-to-end, in the uplink the server measures the data received from the mobile and sends back the results. In the downlink the measurements are made by the same software that generated the uplink data. Quality of service (QoS) parameters Whatever parameters we measure, the ultimate objective at the application layer is to get the user perspective. Thus it is essential to benchmark performance broadly against certain standards. ETSI GPRS recommendations define quality of service for users according to specific parameters, including reliability, throughput, and delay. Our data performance measurements therefore focus on these parameters.
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Reliability measurements Reliability has been defined as the probability of service data units (SDUs) getting corrupted, lost, duplicated, and received out-of-sequence. Service data units are IP datagrams at the application layer. ETSI has defined five classes of reliability at different interfaces. Since our application-layer measurements are made from endto-end, these reliability classes are significant. If we know the negotiated reliability class of the phone, for example, we can make error performance measurements, correlate them with the assigned reliability class, and benchmark performance against the standard figures. Similarly, by performing absolute data-performance measurements, we can determine the highest achievable class of reliability in the network and based on the measurements assign reliability classes to customers. Because SDU reliability is affected not just by errors but also by missequence and loss of SDUs, reliability measurements also help us optimize routing paths, packet control parameters such as segmentation and compression, and system timers for controlling buffer overflow. The table in Figure 7.4 shows the level of reliability desired for the five reliability classes defined by ETSI. Each class defines the interfaces on which protection and acknowledgments are mandatory.
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ECE For non-real time traffic, the highest reliability is required because the application
layer cannot handle data corruption, loss, and other such problems. If traffic is real time, the application layer can manage retransmissions and thus the lower layers require less reliability. Typically reliability classes 2 4 are preferred, with class 2 the ideal. The reason for this preference is that we can manage physical link reliability (quality) on the Gn interface and so ensure reliability on the GTP interface, which is generally in UDP mode. This helps ensure that retransmissions will not occur and thus provides a better throughput. The LLC layer data transfer goes through the air interface as does the RLC. The probability of data corruption is low on the LLC layer if the RLC is reliable, but the probability of missequencing is high as a high degree of segmentation takes place at the LLC and RLC layers. With different levels of reliability possible, the application layer must be able to detect errors when it operates at a higher reliability class. If errors are not corrected by the GPRS layers, then they should be by the application layer. For test purposes, we cannot provide protection at the application layer, so we cannot make the corresponding error performance measurements at this layer. Rather, errors must be measured using UDP on the IP layer at the data reception and generation ends. Quantifying reliability ETSI has attempted to quantify reliability based on the probability of SDUs being lost, duplicated, out of sequence, or corrupted. The SDUs that get corrupted by errors on the interfaces can be delivered to the end application and thus to users in that same corrupted form. Due to problems on the interfaces and in the retransmission process, SDUs also may be duplicated or arrive at the end node out of sequence. GPRS is not a store-and-forward service. Customers are real time data users, so GPRS provides buffers to store information along the route to compensate for resource shortages. Delays in getting resources or transporting the information further depend on the protocols and equipment being used. The SDUs stored in the buffer may be discarded by a GPRS node if the holding timer expires; if this happens these SDUs will be lost. Thus, to meet a specified level of reliability, we must overcome a number of measurement challenges.
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Reliability measurement challenge: Synchronization One of the biggest challenges in measuring reliability is time synchronization. With an end-to-end measurement approach, we need to synchronize the time that the data is sent with the time it is received. In the downlink, synchronizing measurements is generally not a problem. But when the datagrams are sent in the uplink, they reach the measurement server only after some period of time has elapsed. The server then must send the results back to the MS, which adds more delay. As a result, the time at which the data was sent and the time at which the results are received can differ significantly. The position of the mobile may also have changed at this point and thus the serving cell. We can resolve this problem in a couple of ways: We can synchronize the absolute timing at the MS and the server (for example, using the Global Positioning System). The MS can then timestamp every datagram. The server then returns the datagram with the MS time- stamp and adds its own. The timestamps can be correlated to the right position. The MS also can stamp every datagram sent in the uplink with the latitude and longitude. The same latitude and longitude is returned by the serve in the measurement results, providing the correct positions. If there is no synchronization, we can reduce the SDU size, which speeds the transmission of measurement results from the server back to the MS. We can also reduce the LLC block size. Both of these actions reduce the time it takes to transmit results back to the MS allowing results to be much faster.
Figure 7.5 Differences in the time data is sent and measurement results are received create a need for synchronization. Since GGSNs are located indoors, it is important to locate the GPS antenna outdoors in view of GPS satellites.
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Reliability measurement challenge: SDU size When we make data performance measurements, another challenge is to decide the SDU size. In practice, SDU size depends on the application in use. If we look at different layers in the protocol stack, we see the packet sizes that exist in GPRS. The size of an IP datagram can range from 1 octet to 65,535 octets. When a datagram arrives at the SNDCP/LLC layer at the SGSN, it is segmented into 1520 octets. There are other possibilities for controlling packet size in a general way. ETSI defines two SDU sizes for reliability measurements: 128 octets and 1024 octets. Whether we use the smaller or larger size will affect our measurements. If SDUs are small in size, we have to send a larger number of them. Even if a few of them get corrupted, we will still
achieve good reliability percentages. On the other hand, if we use the larger size for our SDUs, we will send fewer of them. In this case, even if only a small number of SDUs get corrupted, the reliability percentage for our data transmission will drop. Further, retransmission of large SDUs adds delay and thus reduces throughput. On the other hand, the larger SDUs require less overhead and so help increase application layer throughput. An appropriate SDU reliability test involves two steps: Simulate SDU performance We can simulate IP data- grams based on different possible applications, measure their reliability, and correlate the results with our throughput measurements. Measure raw bit errors Measuring the raw bit errors gives us an indication of link (MS to GGSN) reliability. Then, correlating the raw bit error measurements with simulated SDUs of different size provides a good estimate of the reliability of the link for different applications. Agilent uses this two-step process to measure IP layer BER performance. Throughput factors One major performance parameter that is evident to customers is throughput. Throughput is the rate at which data is expected or received over a period of time. Throughput is measured in bits/second. Simply stated, it is the data rate achieved against expectations. A combination of several factors determines throughput link reliability, compression, retransmission mechanisms, and delay.
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Link reliability Errors in data transmission at the link can trigger retransmissions that is, the same data being transmit- ted again. The result is a reduction in throughput. Throughput can be defined as either raw or effective. Raw throughput indicates the rate at which data is received. The data may contain errors and therefore would not be usable. The IP layer throughput is generally considered the raw throughput. Effective throughput is the rate of correctly received data. This is generally associated with the application layer. Compression This technique is deployed at the SNDCP layer at the SGSN to reduce the size of the data. Compression increases throughput on the air interface layer, which cannot offer high throughput due to physical layer limitations. Compression affects our measurements at the application layer. Although it increases throughput on the air interface, it may not do the same at the application layer, because the SNDCP has to decompress the data before passing it to the application layer, where limitations on the mobile phone's buffering Retransmission mechanisms These mechanisms are used to compensate for data impairments. But retransmissions are optional and depend on the negotiated QoS class. If retransmissions are on, then SDU size plays an important role in determining the throughput. If retransmissions are not on, then GPRS layer throughput will be high, but if errors are present, the application layer throughput will suffer. In this case, since the lower layers are not compensating for errors, the application layer must. Another important consideration is the air interface cell reselection done by the MS when it is in the packet transfer mode. Cell reselection results in a complete transfer of TBF (temporary block flow) rather than segmented LLC blocks. With many reselections there will be more TBF level retransmissions, which again impact our throughput capabilities may exist. Delay There are several causes of delay in packet transmission and reception. Delays can be caused by buffering. Because the interface types are different and each inter- face has a
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excessive, however, buffers may overflow and SDUs will be lost. This triggers retransmission of the lost packets, which affects the throughput. Delay can also occur if acknowledgements are not received within a specified period (established by timers), again triggering retransmission. Peak throughput Defined by ETSI, peak throughput is measured at the Gi and R reference points in units of octets per second. It specifies the maximum rate at which we expect data to be transferred across the network for an individual PDP context. There is no guarantee that this peak rate can be achieved or sustained for any time period; sustaining peak throughput depends on the capability of the MS and the availability of radio resources. The network may limit customers to a negotiated peak data rate, even if additional transmission capacity is available. The peak throughput is independent of the delay class, which determines the per-packet GPRS network transit delay. The peak throughput classes are defined in Figure 7.6
Figure7.6. ETSI peak throughput classes Measuring throughput Throughput is measured at the application layer and at the GPRS layers. Throughput is affected by factors such as compression and reliability classes, which can create significant differences at the application layer and the GPRS layers. Assigning a high reliability class to data will ensure that throughput at the GPRS layers is the same as throughput at the application layer, as error detection and retransmission will be
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layers and lower throughput at the application layers, as errored SDUs can be passed to the application layer, which then must retransmit the data. Additionally, if the application layer is working in acknowledged mode, the estimated throughput rate will be accurate, but the actual throughput will be lower. This mode requires using TCP at the application layer. With TCP, the probability of detecting errors at the application layer goes down because only error-free data gets through. So, although we get a correct estimate of throughput, we compromise our reliability measurements. A more suitable approach to achieving throughput is to keep the application layer in unacknowledged mode (thus using UDP). This scenario gives us high data rate and the ability to measure errors. By taking into account the number of errors versus the number of data packets received, we can calculate the application layer throughput. This level of measurement flexibility is possible only by making end-to-end measurements. End-to-end throughput measurements End-to-end throughput measurements are made at the IP layer and the application layers. IP Layer Throughput This measurement requires the use of UDP. Since UDP does not manage reliability, all the data coming in at the IP layer is transferred to the application layer. Recall that TCP and UDP are in the layer between the IP and application layers. The application layer is a test client, so we can measure the data coming in from the IP layer that is, the IP throughput. At this stage we don't know how reliable the data coming in is, so the IP throughput measurement can include bad packets. Application layer throughput This measurement can be done in both TCP and UDP modes. When TCP is used, reliability is included in the throughput measurement at the application layer. If UDP is used (which allows us to measure the IP layer performance, as discussed in the previous section), then the application layer throughput is calculated using the IP layer throughput and the bad or errored data at the application layer (that is, the measured IP BER). In UDP mode, we therefore get both IP layer and application layer throughput information, as illustrated in Figure 7.7. In TCP mode, we measure only the application layer throughput.
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Figure7.7. End-to-end throughput measurement at the IP and application layers. IP BER, lost packets, and out-of-order packets are also shown. Delay measurements GPRS is not a store and forward service. It buffers data only temporarily during transmission resource assignment and handling of impairments. Data buffering can result in delay. Delay can vary with time and load. Delay performance is based on the mean delay/second for data transfer through the GPRS network. High rates of delay can occur during momentary problems, such as a moment of peak traffic. To account for this, ETSI defines a 95-percentile delay, which is the maximum allowable delay for 95 percent of the SDUs that are delivered over the GPRS network. The delay parameter defines the end-to-end transfer delay incurred in the transmission of SDUs through the GPRS network. It is measured from mobile data terminal interface (R) to the Gi interface at the SGSN, as shown in Figure 7.8
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synchronize the absolute timing at the transmit and receive ends. This can be achieved by using GPS receivers at both ends or by using some other proprietary time synchronization technique. Once the clocks synchronize, the transmit end ill timestamp the SDU and send it to the receive end. The receive end captures the SDU, adds its timestamp, reads the attached transmit end timestamp, and measures the delay (which is the time that has elapsed between the transmit and the receive timestamps). ETSI defines delay measurements from R at the mobile to the Gi interface. If we want to benchmark against ETSI standards, the measurement server needs to be at the GGSN, within the firewall of the GPRS network (Figure 7. 9). We can also locate the server in the PDN to identify PDN nodes that may be causing excessive delay.
Round trip time (RTT) Since end-to-end delay requires time synchronization, which becomes difficult in certain implementations, RTT is another method of getting some information about the delay. The RTT is the amount of time measured from the point in time at which the last bit of the data packet leaves the application to the point in time at which the last bit of the acknowledgement is
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understanding the impact on data performance of packet fragmentation and buffer size. A combined analysis on both RTT and one way delay allows us to troubleshoot problems of latency in the network. QoS simulation Simulation capability is another critical element of data performance measurements. Customers are assigned QoS levels for different classes of service when they subscribe to a GPRS service. These parameters are also negotiated during PDP context activation. Data performance among QoS levels will vary significantly. If the GPRS network can achieve high QoS levels, those levels can be assigned to customers for a higher price. To verify the data performance, we need the ability to simulate multiple data transactions at the different QoS levels. Because QoS levels are associated with measurement parameters at the GPRS layers, we cannot simulate data
transfer patterns at the application layer. Therefore, we require multiple handsets (or different SIM cards) to make data calls at the different QoS levels simultaneously. Our multi-handset test model for measuring QoS levels should include a data performance test suite that can control the multiple phones and generate similar data-transfer patterns on each phone. With each phone assigned a different QoS level, the performance can be verified at each level and appropriate steps taken to correct any problems.
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the network. Calls can drop in the network due to quality degradation, which may be due to many factors such as capacity limitations, interference, unfavorable propagation conditions, blocking, etc. The DCR is related to the call success rate (CSR) and the handover success rate. The CSR indicates the proportion of calls that were completed after being generated, while the handover rate indicates the quality of the mobility management/RRM in the radio network. KPIs can be subdivided according to the areas of functioning, such as area level, cell level (including the adjacent level), and TRX level. Area-level KPIs can include SDCCH requests, the dropped SDCCH total, dropped SDCCH A-bis failures, outgoing MSC control handover (HO) attempts, outgoing BSC control HO attempts, intra-cell HO attempts, etc. Cell-level KPIs may include SDCCH traffic BH (av.), SDCCH blocking BH (av.), dropped SDCCH total and distribution per cause, UL quality level distribution, DL quality/level distribution etc. The TRX level includes the likes of UL and DL quality distribution.
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8.3.1 COVERAGE Drive test results will give the penetration level of signals in different regions of the network. These results can then be compared with the plans made before the network launch. In urban areas, coverage is generally found to be less at the farthest parts of the network, in the areas behind high buildings and inside buildings. These issues become serious when important areas and buildings are not having the desired level of signal even when care has been taken during the network-planning phase. This leads to an immediate scrutiny of the antenna locations, heights and tilt. The problems are usually sorted out by moving the antenna locations and altering the tilting of the antennas. If optimization is being done after a long time, new sites can also be added. Coverage also becomes critical in rural areas, where the capacity of the cell sites is already low. Populated areas and highways usually constitute the regions that should have the desired level of coverage. A factor that may lower the signal level could be propagation conditions, so study of link budget calculations along with the terrain profile becomes a critical part of the rural optimization. For highway coverage, additions of new sites may be one of the solutions. 8.3.2 CAPACITY Data collected from the network management system is usually used to assess the capacity of the network. As coverage and capacity are interrelated, data collected from drive tests is also used for capacity assessment. The two aspects of this assessment are dropped calls and
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congestion. Generally, capacity-related problems arise when the network optimization is taking place after a long period of time. Radio network optimization also includes providing new capacity to new hot-spots, or enhancing indoor coverage. Once the regional/area coverage is planned and executed in the normal planning phase, optimization should take into consideration the provision of as much coverage as possible to the places that would expect high traffic, such as inside office buildings, inside shopping malls, tunnels, etc. 8.3.3 QUALITY The quality of the radio network is dependent on its coverage, capacity and frequency allocation. Most of the severe problems in a radio network can be attributed to signal interference. For uplink quality, BER statistics are used, and for downlink FER statistics are used. When interference exists in the network; the source has to be found out. The entire frequency plan is checked again to determine whether the source is internal or external. The problems may be caused by flaws in the frequency plan, in the configuration plans (e.g. antenna tilts), inaccurate correction factors used in propagation models, etc.
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quality exceeds 6, then the networks performance is not acceptable. It can be found out through a long call as well as a short call. C/I ratio: The C/I ratio or the Carrier to Interference ratio is an important parameter in analyzing the quality of a network. The C/I ratio gives a relationship between the carrier of the network and the interference it is facing. Theoretically the C/I ratio should be at least 18 db but in practical cases 12 db is also acceptable. Handover success rate (HO Rate): HO rate is the number of successful handovers made by a cell phone. It is the ratio of the number of successful handovers made to the total number of attempts to make a handover. To find out the HO rate, the mobile should be in dedicated mode i.e. the mobile should be on call. HO rate = number of successful handover attempts/ total number of handover attempts. Through put : the speed of the data that is received and transmitted by the network at that particular location. It is calculated by dividing the total average speed to the no.of throughputs. Apart from these parameters the following parameters are also considered: Frame Error Rate; Cell Site Database- Site Configuration, Latitude & Longitude of the site location, BSIC, LAC, Hopping Frequencies, Non-Hopping Frequencies; MAIO, Antenna Parameters like Tilt, Pattern, Gain, and Azimuth/Orientation.
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Figure 9.1 Integrated drive-test solutions consisting of a digital receiver and phone. A GPS receiver provides location information.
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A notebook - or other similar device (1) equipment. The following is a schematic of the standard connections. With collecting Software installed (2), A Security Key - Dongle - common to these types of software (3), At least one Mobile Phone (4), One GPS (5), A Scanner optional (6).
Also is common the use of adapters and / or hubs that allow the correct interconnection of all
The main goal is to collect test data, but they can be viewed / analyzed in real time (Live) during the test, allowing a view of network performance on the field. Data from all units are grouped by collection software and stored in one or more output files (1).
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GPS: collecting the data of latitude and longitude of each point / measurement data, time, speed, etc. It is also useful as a guide for following the correct routes. MS: mobile data collection, such as signal strength, best server, etc ... SCANNER: collecting data throughout the network, since the mobile radio is a
limited and does not handle all the necessary data for a more complete analysis. The minimum required to conduct a drive test, simplifying, is a mobile device with software to collect data and a GPS. Currently, there are already cell phones that do everything. They have a GPS, as well as a collection of specific software. They are very practical, but are still quite expensive.
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Some software allows the image to be loaded as the software background (georeferenced). This makes it much easier to direct routes to be followed. It is advisable to check traffic conditions by tracing out the exact pathways through which the driver must pass. It is clear that the movement of vehicles is always subject to unforeseen events, such as congestion, interdicted roads, etc. Therefore, one should always have on hand know - alternate routes to be taken on these occasions. Avoid running the same roads multiple times during a Drive Test (use the Pause if needed). A route with several passages in the same way is more difficult to interpret.
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Short calls should last the average of a user call - a good reference value is 180 seconds. Serve to check whether the calls are being established and successfully completed (being a good way to also check the network setup time).
Long calls serve to verify if the handovers (continuity between the cells) of the network are working, i.e. calls must not drop.
Tests for Analysis Performance is the most common, and usually made into clusters (grouping of cells), i.e., an area with some sites of interest. They can also be performed in specific situations, as to answer a customer complaint. In integration testing of new sites, it is recommended to perform two tests: one with the site without handover permission - not being able to handover to another site thus obtaining a total visualization of the coverage area. The other, later, with normal handover, which is the final state of the site. Depending on the type of alteration of the site (if any change in EIRP) both tests are also recommended. Otherwise, just perform the normal test. Marketing tests are usually requested by the marketing area of the company, for example showing the coverage along a highway, or at a specific region/location. Benchmarking tests aims to compare the competing networks. If the result is better, can be used as an argument for new sales. If worse, it shows the points where the network should be improved.
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Also, the following data from the OMC-R is to be collected: BCCH frequency Hopping Frequency MAIO & HSN Neighbor List
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The drive test is done over a distance of 3 km or more from the starting point. Various parameters are observed and recorded during the drive test.
The drive test procedure is as follows: Tool may be setup for two mobiles One for Long call and another for short calls (2 minutes). In the route map following are to be enabled for Analysis. Rx Level RX Quality Survey Markers (like H/O, DCR & H/O symbols) Cell site Database.
Call statistics for the Calls in the Point -1 to be enabled. Conduct the Drive Test covering all sectors by observing the following Parameters: Rx Level Rx Quality Interference on BCCH & Hopping Frequencies. Handovers & Drop Calls Observe whether the nearest sector is serving or not.
The data, as per the requirements are observed and recorded. The data is analyzed for performance.
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File
One mobile for short call configuration One mobile for Long call configuration GPS ProjectManager NewProject Name
ViewSystem panelsHard ware window(right click)Add DevicePhone(short call) ViewSystem panelsHard ware window(right click)Add DevicePhone(Long call) ViewSystem panelsHard ware window(right click)Add DeviceGPS
Figure 10.1 Configuration of Hard ware Devices. 10.3.2 CONFIGURING THE CALLS In Sequencer window we specify the type of test to be done by the each device i.e. mobiles For Short Call:
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ViewSystem panelSequencer (right click)Service model (right click)Parallel sequenceShort call For Long Call: ViewSystem panelSequencer (right click)Service model (right click)Parallel sequenceLong call
Figure 10.2 Configurations of Calls 10.3.3 CONFIGURING SHORT CALL PROPERTIES Short CallCALL_CONTROL_TESTViewProperties Number of times to run: Infinite -The number of times for a call to run throughout the Drive Test if after a disruption. Inter Call Idle time: 5 sec -Time duration between the calls Auto Dial: Yes -Makes the call automatically after 5 sec (Inter Call Idle time) Call Statistics: Yes -Display of No. of Dropped calls, Good calls etc., (Call Analysis)
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Continuous Call: No -Since the Short Call would be terminated and re-initiated throughout the Drive Test, it is configured as No
Call Duration: 40 sec -Duration of the Short Call should be minimum to make a trail for every sector or cell
Call Setup: 20 sec -Time given to setup or answer a call, if it exceeds call will be terminated.
Auto Answer: No -If it is Yes, then the mobile would be only in incoming mode (doesnt suit for Drive Test)
Voice MOS Test: No -It is the Voice Mean Opinion score Test, not required for the Drive Test because person doing the Drive Test dont speak throughout the Test.
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Figure 10.3 Configurations of Short Call Properties 10.3.4 CONFIGURING LONG CALLS PROPERTIES Long CallCALL_CONTROL_TESTViewProperties Number of times to run: Infinite -The number of times for a call to run throughout the Drive Test if after a disruption. Inter Call Idle time: 5 sec -Time duration between the calls Auto Dial: Yes -Makes the call automatically after 5 sec (Inter Call Idle time) Call Statistics: Yes -Display of No. of Dropped calls, Good calls etc.,(Call Analysis) Immediate Dial: Yes -To dial immediately after disconnection. Continuous Call: Yes -Since the Long Call would be operated throughout the Drive Test, it is configured as Yes Call Duration: NILL
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-As the Long Call is operated throughout the Drive Test the Call duration need not be specified. Call Setup: 20 sec -Time given to setup or answer a call, if it exceeds call will be terminated. Call number: Any number -Destination or called party number Auto Answer: No -If it is Yes, then the mobile would be only in incoming mode (doesnt suit for Drive Test) COM Port: COM 57 -Number of port that to be connected to PC Voice MOS Test: No -It is the Voice Mean Opinion score Test, not required for the Drive Test because person doing the Drive Test dont speak throughout the Test.
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Figure 10.4 Configurations of Long Call Properties 10.3.5 CONFIGURING OF MAP AND CELL SITE DATA MAP: ViewCommon ViewsMapOpen map file (Load from Destination address in PC) Cell site Data: ToolsOptionsCell siteOpen cell site data file (Load from Destination address in PC) Hyderabad map like streets, state highways, water bodies, national highways etc with cell sites given below
Figure 10.5 Cell Sites of Gachibowli. 10.3.6 CONFIGURING THE DATA ITEMS Configuring the Data Items Selection of parameters like Rx level, Rx Quality, C\I ratio of both Short call and Long call which we want to display in the map through different colors and different ranges which are available in Data items window.
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For Short Call: ViewSystem PanelData itemsShort CallTraceGSM signal -Rx Level -Rx Quality
For Long Call: ViewSystem PanelData itemsLong CallTraceGSM signal -Rx Level -Rx Quality -C\I Ratio
10.3.7 MAP LEGEND Map Legend shows the display of Configured Data Items Selected like Rx level, Rx Quality, C\I ratio of both Short call and Long call in the map through different colors and different ranges as shown in the Legend Window in the below figure.
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GPS location or Vehicle position on the map is indicated with red pointer as shown below. Five parameters like Rx quality, Rx level of short and long call and C/I ration of long call shown below on the map.
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BSNL drive test from RTTC Gachibowli to Nanakramguda , Khajaguda via Outer ring road and back to RTTC via Cyberabad Commissioner office. In the map red color indicates the bad quality. A map legend (Indications of 5 parameters and their ranges) is shown on right side of screen.
BSNL user events like H.O success, good call etc can be seen below.
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11.2.2 FORMULATION & CALCULATION (1) Rx Level: Rx Level = Total samples (> -95 dbm) Total no. of samples collected (a) BSNL: Total no. of samples collected= 5225 Total no. of samples collected (> -95 dbm) =5101 Rx Level = 5101 --------- *100 % 5225 = 97.62% *100 %
(2) Rx Quality:
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Rx Quality=
(a)
BSNL: Total no. of samples collected= 5225 Total no. of samples collected (=0) =264 Total no. of samples collected (=1) =84 Total no. of samples collected (=2) =199 Total no. of samples collected (=3) =362 Total no. of samples collected (=4) =2008 Total no. of samples collected (=5) =1553
Total no. of samples collected (0-5) =264+84+199+362+2008+1553 =4470 Rx Quality= 4470 --------5225 *100 % = 85.5%
(3) Carrier to Interference Ratio (C\I): C\I = Total samples (> 9 db) Total no. of samples collected BSNL: Total no. of samples collected= 3990 Total no. of samples collected (> 9 db) =3863 *100 %
C\I
3863
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-------3990
*100 %
(4) Hand Over Success Rate: HSR = Total no. of successful H.Os Total no. of H.O commands BSNL: Total no. of Hand Over Commands = 52 Total no. of Successful Hand Overs=50 *100 %
HSR
(5) Call Analysis: (i) Call Setup Success Rate: Rate of calls which are successfully established. CSSR= No. of calls successfully setup Total no. of calls attempted * 100
(ii)
Call Completion Success Rate: Rate of calls which are successfully established and disconnected by the user. CCSR= No. of calls setup and disconnected * 100 Total no. of calls Attempted
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(iii)
Drop Call Rate: DCR= No. of Dropped calls Total no. of calls Established S.no 1 2 3 4 (a) Parameter No of Call attempts Successfully Established No. of Blocked calls No. of Dropped Calls BSNL: (i) CSSR = 20 21 (ii) CCSR = 20 21 (iii) DCR = 0 20 * 100 = 0% * 100 = 95.2% * 100 = 95.2% BSNL 21 20 1 0 * 100
(6) THROUGHPUT : max bit rate of data transmitted BSNL = 70 KBPS (7) SERVICE DELAY : BSNL = 7.04MS (8) FER (FRAME ERROR RATE ) BSNL = 5150 5220 * 100 = 98.6%
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12.1 ADVANTAGES
Operators would able to solve the problems of network. We can identify and compare the performance of different operators. Operators can provide better service to the customers.
12.2 DISADVANTAGES
Expensive. Time taking process. Operator has to engage engineers to collect the data (Bench marking).
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CHAPTER 13 APPLICATIONS
For every operator, customers are very important. Before customer complaints (like call drops ,bad speech quality, no signal etc.),the operator should able to identify the problem and rectify the problem i.e. preventive maintenance . corrective maintenance i.e. After receiving complaint from the customer, operator should solve the problem.
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At present Drive Testing in GSM RF Optimization is being performing manually for the improvement of performance of the network. Instead of doing drive testing manually, there may be a scope ANMS (Automatic Network Management System) process. In this system, Drive Testing equipment can be attached to moving vehicle to serve in operator test area and it can be monitored by the server. By using the internet, all the drive data can be simultaneously collected up to date to the server.
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CHAPTER 15 RESULT
COMPARISION OF BENCH MARKS
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CHAPTER 16 CONCLUSION
The overall objectives of any RF design depend on a number of factors that are determined by the needs and expectations of the customer and the resources made available to the customer. Due to the mobility of subscribers and complexity of the radio wave propagation ,most of the network problems are caused by increasing subscribers and the changing environment. Radio Network Optimization is a continuous process that is required as the network evolves. Radio Network optimization is carried out in order to improve the network performance with the existing resources. The main purpose is to increase the utilization of the network resources, solve the existing and potential problems on the network and identify the probable solutions for future network planning. Through Radio Network Optimization, the service quality and resources usage of the network are greatly improved and the balance among coverage, capacity and quality is achieved.
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CHAPTER 17 BIBILOGRAPHY
1. Wireless communication by S.Rappaport 2. Mobile cellular telecommunication - William C.Y.Lee 3. Cellular technology for rural areas W.C.Y.Lee 4. Umts performance Measurement by Ralf Kreher 5. GSM, GPRS and EDGE performance by Timo Halonen,Javier Romero,Julan Melero 6. GSM/EDGE: Evolution and Performance by Mikko Saily,Guillaume Sebire, Eddie Riddington
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CHAPTER 18 REFERENCES
1. http://en.wikipedia.org/wiki/GSM 2. www.jdsu.com 3. http://www.telecombuzz.org/2009/08/gsm-drive-test.html 4. www.fcc.gov (documentation from Motorola) 5. www.ofcom.org.uk 6. www.etsi.org (definitions of GPRS)
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ABBREVIATIONS
2G Second Generation AB:-Access Burst ANSI American National Standards Institute Auk Authentication Centre BSC Base Station Controller BSS Base Station System BTS Base Transceiver Station CC Call Control CDMA Code Division Multiple Access CM Connection Management DCS 1800 Digital Cellular System 1800 (today: GSM1800) DECT Digital Enhanced Telecommunications System DL Down-Link DRX Discontinuous reception DTX Discontinuous Transmission EDGE Enhanced Data rate for GSM Evolution EGPRS Enhanced General Packet Radio Service EIR Equipment Identity Register ETSI European Telecommunications Standards Institute FDMA Frequency Division Multiple Access GMSC Gateway MSC GMSK Gaussian Minimum Shift Keying GPRS General Packet Radio Service GSM Global System for Mobile Communications HLR Home Location Register
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HSCSD High Speed Circuit Switched Data IMEI International Mobile Equipment Identity IMSI International Mobile Subscriber Identity IP Internet Protocol ISDN Integrated Services Digital Network Kbits/s/slot Kilo Bits per second per slot Kb Kilo bits KB Kilo Bytes ME Mobile Equipment MM Mobility Management MS Mobile Station MSC Mobile Switching Centre MSRN Mobile Station Roaming Number OMC Operation and Maintenance Centre PDP Packet Data Protocol PLMN Public Land Mobile Network PSK Phase Shift Keying QoS Quality of Service SCH Synchronization Channel SIM Subscriber Identity Module SMS Short Message Service SMSS Short Message Service Support SS Supplementary Service Support SS7 Signalling System Number 7 TD/CDMA Time Division Code Division Multiple Access TDMA Time Division Multiple Access TMSI Temporary Mobile Subscriber Identity
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TRAU Transcoding and Rate Adaptation Unit TRX Transceiver UE UT User Equipment User Terminal UL Up-Link UMTS Universal Mobile Telecommunications System USB Universal Serial VLR Visitor Location Register
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