In order to satisfy constantly increasing user demands, in process of transition towards LTE (Long Term Evolution) mobile networks of 4th generation, for 3G UMTS networks potential solution is exploitation of multi-antenna MIMO (Multiple Input Multiple Output) systems, which provide increase in both capacity and reliability. This paper puts an effort to provide a link between link-level and system-level gains introduced by multi-antenna techniques in UMTS systems. Coverage, capacity and achievable data rates are evaluated and presented for Alamouti scheme, TxAA and D-TxAA. Radio mobile planning tool is used for illustration of coverage improvement achieved by implementation of several multi-antenna techniques. Considering system-level gains, it is seen that implementation of MIMO techniques yields larger service area and/or increase in capacity and available data rates maintaining the same bandwidth. Results presented in this paper help making a clear picture of achievable gains, employment costs and answering the question: Should mobile operators invest in upgrading their 3G networks with MIMO techniques and is that investment justified, considering achievable improvements.

—In this paper, we propose an algorithm for adaptation layer in order to improve fairness in bandwidth allocation among different traffic classes in IP/MPLS networks under heavy traffic load. We propose a definition of the blocking frequency of traffic flows at the entry of autonomous network domain and proportional-priority coefficient per traffic class and use them as the input parameters of the adaptation mechanism. In order to evaluate the validity of the proposed algorithm, we need proper simulation tools and for that purposes we extend OPNET Modeler 14.5 with the modules for adaptation process. We also propose a development methodology for the design of modules for adaptation process within network simulators. The simulations results give us the proof of our hypothesis that with a proper adaptation layer we can improve the fairness of bandwidth allocation among different traffic classes under heavy network load and at the same time keep the required QoS conditions in the preferred boundaries.

— This paper gives a proposal on algorithm for adaptive adjustment of VoIP sources transmission rate based on voice quality estimated at the receiver. This adjustment is achieved through the appropriate use of differing voice codecs, as changes in network conditions occur, in order to maintain an efficient utilization of the available resources. We propose a simple algorithm for dynamic selection of voice codec, depending on network conditions during the on-going voice session. Algorithm is embedded in the source code of the programming environment OPNET Modeler 14.5. Simulation results show that the proposed algorithm makes better use of the available bandwidth, achieving superior performance in comparison to the situation without implementation of algorithm for adaptive codec selection.

Currently, among the main three affectors of transmission quality (represented by the mean probability of a bit error) for a particular digital modulation format, such as eg. GMSK, fast fading seems to be by far mostly dealt with, in particular, analytically, while such an analysis for the other two aspects, namely, slow fading and quasideterministic attenuation of the signal is usually too complex, often resulting with analytic solutions of mainly academic relevance, rather than practical, while numerical and/or simulation methods are very time-consuming and not explicitly conclusive. Therefore, in this paper, we focus simultaneous fast Nakagamian and slow (shadowing) fading, providing the explicit analytical expression for the probability density function of a mobile channel signal composite envelope, and consequently the according assessment of the transmission performance, that is finally supported by extensive computer experiments for characteristic parameter values of fast and slow fading.