Recent advances in both satellite and terrestrial mobile communications technologies are now leading to the realisation of the dream of the global personal communications within a few years. Satellite systems, as a complement to terrestrial cellular systems, are introduced into the future Personal Communication Networks (S-PCN) to provide global coverage and to allow global roaming. The inter-working and the integration between the satellite and the terrestrial cellular systems (e.g. GSM system) are the key issues in developing the network architecture and designing the control functions and signalling protocols of satellite systems. This thesis focuses on the design of a satellite signalling control system. The coverage and link properties of ICO10 and LE066 satellite constellations, the representatives of low earth orbit (LEO) and medium earth orbit (MEO) satellite systems, are considered. A satellite specific network architecture is proposed to accommodate the requirements of satellite dynamics and resource control function. The physical layer of satellite signalling links are designed to cope with the specific features of LEO or MEO satellite air-interfaces. In order to overcome problems specific to LEO or MEO satellite systems and to provide call set-up control function, three important signalling protocols are proposed for the S-PCNs. The priority based fast access scheme is designed for the satellite random access channel allowing low access delay for the call set-up related access packets, even when the channel load is high. The satellite diversity based paging approach is proposed to optimize the paging performance. The modified selective re-transmission (M-SRT) and Go-Back-N (M-GBN) protocols are proposed to cope with the transaction type transmission on the dedicated control channel. Simulation results have shown siginficant improvement of the M-SRT and M-GBN protocols in call setup delay. Two protocols are also compared in the aspects of implementation complexity and call set-up performance. Finally, the integration scenarios between satellite and GSM system have been examined for S-PCN in the call handling related functions and associated signalling protocols. The GSM higher layer signalling protocols are tailored to provide the call control related functions. The optimum integration scenario is derived under the criterion of minimum modifications to the GSM higher layer signalling protocols and minimum complexities of the control functions.

Several recent studies show that network traffic is self similar, or exhibits long range dependent characteristics. Self similar traffic is problematic for routing and congestion control algorithms because self similar traffic is very different from conventionally considered traffic such as Poisson or Markovian traffic. Self similar behaviour is expected to occur in future data networks as well as can be seen in present networks carrying bursty services. Therefore self similar behaviour must be thoroughly understood if appropriate call admission controls, scheduling algorithms and congestion control mechanisms are to be designed. Also characteristics of data traffic play a crucial role in performance analysis and design of communication networks. Understanding the models of network traffic helps designing better protocols, better network topologies, better routing and switching hardware and provide better services to the users. Therefore the need of traffic characterisation is a major challenge faced by network engineers at present. This research illustrates the different service modelling distributions significantly changes the medium access control performance. This is validated against two popular service models for WWW browsing and Email connections. WWW browsing is modelled using heavy tailed Pareto distributed burst sizes characterizing self-similarity at the aggregate traffic level. Email sessions are presented with the Cauchy distributed connection sizes. The results conclude the different service modelling distributions have a significant impact on medium access control performances. Investigation of call admission control and scheduling algorithms for diverse service classes is also studied in this work. A novel admission control and scheduling algorithm-based on evolutionary algorithms is proposed and the superior performance of the proposed technique over the state of the art mechanisms is demonstrated on an example GPRS system.

In recent years the rapid growth in the number of telephone users world-wide, particularly in the area of mobile communications, has highlighted the Quality of Service offered by the service providers as an increasingly important issue. The perceived level of service and the accompanying cost offered to the users will often be a major issue in deciding on which operator to choose for a particular application. Owing to the advantages that digital transmission offers over the original analogue systems, it has become the dominant current/future technology in both the PSTN for trunk telephony, and the emerging PCN and DMR standards, owing to its robustness to channel degradations, signal regeneration, cost, flexibility, increased capacity, switching etc. However, the adoption of digital technology for applications such as satellite and land-mobile systems will require the original speech information to be compressed in order to allow for more available channels within the limited bandwidth allocation. The compressed data will suffer from various levels of distortion caused by the deep fades and multipath signals that are experienced during mobile connections. The effect of these distortions can be reduced by the application of a suitably optimised channel coding and frame substitution strategy. This thesis will present a detailed description of the development and real-time implementation of an 11.4 kbps combined speech and channel coding scheme, that was specifically designed to meet the stringent system constraints of a DMR application. The trade-offs required to allow the entire system to be realised in real-time are discussed, which demonstrated how important it was to include complexity issues in the initial design of any future standards. It also highlights one of the major degradations that may occur when the current and future digital standards are operated in tandem. The problem of tandeming digital voice codecs can cause increased distortions that reduce the overall link quality, owing to limitations in the design of the adaptive predictors which are essential to enable the reduction in coded bit rate. This will have a direct bearing on the quality of service offered by a network operator, as transparent interoperability of the different systems will be expected by the users. The development of a unique system designed to allow the network gain to be measured for a digital mobile to PSTN connection is also discussed. This type of test equipment will be required for future systems as the traditional techniques involving pure tones are no longer applicable. This is because of the assumptions required in the models used by the voice codecs to enable the high levels of compression, whilst still providing the necessary perceptual quality.

In a mobile environment, communication links are subject to a wide range of channel variations due to mobility, multipath fading, shadowing, interference, etc. These variations combined with limited allocated bandwidth often constrain the design of such links and result in degradation of the quality of service and lowered capacity. Techniques such as error correction code are often used to mitigate channel errors at the expense of bandwidth. Other techniques such as the use of smaller cells with extensive frequency re-use are used to enhance system capacity. However, this operation approaches the theoretical channel capacity and increases the likelihood of errors on the channel due to co-channel interference. Although marginal channel conditions are quite common in mobile communication links, they are intermittent. These conditions are encountered for example just prior to handover; during deep shadowing such as is experienced when a communicating mobile suddenly goes behind a building or hill; during the power ramp-on phase of new mobiles joining the system etc. Therefore, the design of a mobile communication link that can adapt to such conditions when it is necessary would be ideal. A multi-rate source and channel coding (MRC) is one form of link adaptation which represents a new approach to achieving consistent high quality speech combined with efficient spectrum usage. In MRC, link adaptation is achieved by trading-off the gross bit rate of each traffic channel between the speech and channel codecs according to the prevailing channel conditions. When channel conditions degrade excessively, the system allocates more bits for error protection thereby operating with a lower bit rate speech codec. In this case, the powerful error protection might well correct all the errors and thus speech synthesis proceeds with uncorrupted parameters. This thesis presents a design of a CDMA-based MRC system. A quad-rate speech coder is designed based on Pulse Residual Excited LPC (PRELP), an algebraic CELP coder. Unequal error protection is used in the design of a quad-rate convolutional channel coder. In order to evaluate the output speech quality and capacity over realistic channel conditions, a CDMA-based comprehensive dynamic link simulator (DLS) is developed and used as a testbed. The simulation results of the MRC show that it is capable of maximising the output speech quality during good channel conditions and maintaining a satisfactory quality in marginal channel conditions. Whilst the lower rate speech codec is expected to produce reduced quality speech, the perceived degradation is shown to be less than that of a fixed rate system in which the error protection would quickly fail, possibly resulting in cessation of the link. Errors on the rate switching commands show no impact on the output speech quality. Indeed, the MRC system keeps its integrity with severe channel conditions and occasional command errors. This is assisted, however, by the careful design of the switching algorithm in which switching can only occur in consecutive modes. Since the multi-rate system permits the use of minimum resources of speech and channel coding bits, capacity of the mobile system is obviously increased. However, in this work it is shown that the multi-rate system can be particularly useful in crowded cells where it can increase the robustness of the system and enhances the capacity.

This thesis covers the application of fade counter measure techniques to satellite links using small terminals in the 20/30 GHz band. The case for using the 20/30 GHz band for very small aperture terminal (VSAT) networks is presented together with applications proposed for VSAT networks. A study of the propagation impairments that affect these networks is presented and the propagation environment is determined with particular reference to the problem and severity of rain fading. Relevant equations are presented allowing the calculation of link budgets and performance for VSAT networks operating via a 20/30 GHz satellite transponder. The various methods that may be employed to detect and overcome the high attenuation levels found at 20/30 GHz are presented. Candidate fade counter measure (FCM) schemes, fade detection mechanisms and control methods are discussed, together with their applicability to VSAT networks and their expected performance. The design of the prototype 20/30 GHz VSAT Earth station developed is presented, together with performance simulations, backed up by measurements, made on the prototype. The application and performance of fade counter measure schemes and fade detection methods is determined by computer simulation. Finally recommendations are made for a complete fade counter measure system applicable to a 20/30 GHz VSAT system.

This thesis begins by addressing the problems and challenges faced in a multimedia, ATM compatible. Wireless LAN environment. A brief overview of the ATM will also be presented. An extension to the conventional wireless (cellular) architecture, which takes advantage of the ATM characteristics, is considered. The needs of the applications that will use such a network along with the services this network is expected to offer, are discussed. The existing wireless protocols (both in the wireless LANs and in the cellular architecture) are presented. The differences among the MAC schemes and discussion on the criteria they must satisfy to support ATM are brought up. Finally, the most promising MAC schemes are discussed in further details. Furthermore, the introduction of terminal and user mobility in an ATM LAN causes the need for modification of the existing network functions. There are new problems, associated with the mobile stations that must be addressed, such as location management, paging, registration, authentication and network security and handover implementation. Furthermore, existing functions of the fixed ATM, such as Connection Admission Control and traffic shaping need to be extended to support the requested QoS in the wireless environment. The next part of the report discusses the different types of handover mechanisms and presents possible extensions of the UNI and PNNI that support the exchange of handover messages. The concept of the Mobile Agent (MA) is also introduced and its use in the extended UNI and PNNI for handover execution, registration and location management purposes is presented. Finally, a further extension of the PNNI protocol, which could be used among the different MAs, in order to support portability across different private ATM LANs, is being discussed. Without a doubt there will be a requirement for interworking between ATM and the already established wireless networks (e.g. HIPERLAN, DECT, IEEE 802.). The use of ATM as a wireless network backbone is particularly advantageous in microcell and/or integrated voice/data scenarios, and is cost-competitive with other possible implementations. Taking that into consideration, the transmission of ATM cells over a WLAN, based on the IEEE 802.11 MAC layer has been investigated. Initially, the IEEE 802.11 MAC layer and its model (developed in BONeS software package), are being discussed. The simulation results show the existence of upper bound delays for delay sensitive applications, such as voice and video, which are not affected by the traffic load on the network. Moreover, a power saving addition to a Dynamic Time Division Multiple Access (D-TDMA) MAC protocol, suitable for ATM cell transmission, discussed in [APOS95], will be proposed. A short presentation of the MAC protocol and the proposed power saving algorithm will follow. The trade-off between the power saving gain and the size of the buffer in the Access Point (AP) are shown for different kinds of services. Finally (based on Markov chains), the calculation of the call blocking and dropping probabilities for different services in a radio environment will be addressed. This method considers both the different QoS requirements for each service and the load on the network. The results obtained with the analysis are being compared to the ones obtained from simulations.

This thesis presents a study of the application of satellite communication techniques for the establishment of Business communication networks. The characteristics of the traffic and its potential market in Western Europe is investigated. This is shown to be of a multiservice nature with the speech being a dominant source of traffic. It is however being indicated that videoconferencing could dominate the traffic mix and overshadow all other services. A Reservation Time Division Multiple Access (R-TDMA) protocol is hence developed to cater for this multiservice traffic. The basic protocol is designed to operate via a transparent satellite transponder. However an extension of the protocol is specifically designed to operate via a regenerative processing transponder. This is a R-TDMA with Free-Slots Contention (R-TDMA/FSC). Both protocol versions are modelled by Numerical Petri-Nets (NPN) and evaluated by computer simulations using the CSIM simulation system. Different network and operating conditions are simulated to test the performance of the protocols. They are shown to offer low delays, high throughputs and stable operation. An analytical model is used to validate these simulation models and results. The design of a second generation business satellite payload incorporating regenerative processing transponders is also carried out as part of the design of the Communications Engineering Research Satellite (CERS). This is shown to dramatically reduce the user earth station complexity and cost. The functional description of such an earth station is given that has the capability of running the earlier developed protocols.

No description available.

This work addresses the problem of digital terrain modelling for estimating radio path propagation within a mobile communication system. The ideal requirements are for a data structure which is storage efficient and computationally efficient for calculating profiles, whilst elevation errors should be constrained and radio path loss errors should be minimised. For a digital terrain model (DTM) to be considered viable as an alternative to the regular grid, it should: (i) produce a storage saving of at least 75% over the regular grid; (ii) be error constrained to a maximum absolute error of 10 metres; (iii) produce only a small overall average elevation error; (iv) preserve critical terrain characteristics such as ridges, peaks and slopes; (v) produce 95% of profiles to within a radio path loss error of ± 6 decibels; and (vi) be as computationally efficient as the regular grid. This research focuses on the implementation of a number of prototype DTMs, including a regular grid, sub-sampled grids, variable density grids, elevation difference grids, polynomial models of fixed and variable degree, surface patch quadtrees, and triangulated irregular networks (TINs). Each of these DTMs are examined in terms of the criteria outlined above. No DTM fulfils all of these requirements. The user should identify the relative importance of each requirement before selecting a specific model. For this study, computational efficiency is identified as the criterion which can be considered the least important. With this in mind, two original DTMs are developed. These are optimised with respect to storage and error constraints. The proposed Huffman-encoded DTM represents the deviations of a regular grid of heights from linearly predicted values as variable-length codes, whilst the Implicit TIN is a storage-efficient triangulated irregular network which reconstructs the original topology of the triangulation at the application stage. Both methods produce storage savings approaching 90% over the regular grid for the data sets tested and are suitable for parallel implementations.

Subcarrier multiplexed (SCM) systems are an attractive alternative to the evolving digital technology for transmitting broadband services, at an affordable price. However, the majority of existing systems are based on analogue signal transmission and therefore, the strict noise and nonlinear requirements undermine the system performance. The work carried out in this thesis presents the feasibility of pulse time modulation (PTM), as a second stage modulator, in SCM systems. PTM techniques offer simplicity and low cost, and with the additional bandwidth available on optical fibres can trade bandwidth to significantly higher signal-to-noise ratio (SNR) levels, compared to analogue systems. Three different PTM techniques, square wave frequency modulation (SWFM), pulse frequency modulation (PFM) and pulse position modulation (PPM) has been investigated. A prototype system capable of transmitting a video channel, two audio channels and a data channel is implemented for each technique in order to evaluate the performance potential of PTM as a second stage modulator in SCM systems. The SNR expressions for all three schemes are derived from the first principles and the obtained results were verified experimentally. The optimum SNR performance is delivered by a raised cosine shaped pulse and the PPM technique delivers 5 dB SNR improvement over PFM. For SWFM systems a 3 dB SNR advantage is gained over single-edge detection technique and PFM systems by employing double-edge detection at the receiver. PPM spectrum contains a clock component which could be employed at the receiver for signal recovery. Demodulation technique, based on clock recovery using a phase locked loop (PLL) is proposed and implemented. This technique is cost effective and less complex compared to the existing demodulation schemes. The PFM implementation shows a 6 dB improvement in the receiver sensitivity compared to conventional SCM systems, while the PPM system offers an extra 2.5 dB improvement. The improved receiver sensitivity of the SCM-PTM technique, results in an increased optical power budget, where the transmission distance, number of subscribers and the number of channels in a network can be optimized. The nonlinear performance of the overall system is also shown to be within the specified performance levels.