This thesis introduces new results in auto-tuning and self-optimization of RRM parameters in 3G and beyond 3G networks. Auto-tuning tasks are organized in the control plane where different information exchange is involved between the network nodes. Auto-tuning using fuzzy logic control is performed in a local loop, namely the controller is in continuous interaction with the network. The controller feeds the network with new parameter settings and conversely the network returns its feedback by delivering new quality indicators indicating its operating state. Different use-cases are investigated. First, an auto-tuning of resource allocation algorithm in UMTS is studied as an alternative to the existing static resource allocation. The auto-tuning process dynamically adapts a guard band that is reserved for users using real time services. A best trade-off between real time and non real time services is achieved in the sense that the quality of service becomes comparable in the two traffic classes especially in a high load situation. The second use-case concerns the self-optimization of soft handover parameters in UMTS networks. For each cell the controller receives as inputs the filtered downlink load and that of its neighbouring cells. The controller continually learns the best parameter values in each network situation. The learning process is governed by a utility function. Simulation results reveal significant improvements in terms of network performance. The proposed auto-tuning algorithm balances the radio load between base stations and improves the system capacity by up to 30% compared to a UMTSnetwork with fixed soft handover parameters. However, the auto- tuning increases the signalling messages load in the radio interface as well as in the core network. This negative effect is minimized by reducing the reactivity of the auto-tuning controller.
Study of WLAN-UMTS hybrid networks is an emerging area of research and not much related work is available. Au- thors in some related papers , , , , , ,  have studied issues such as vertical handover and coupling schemes, integrated architecture layout, radio resource management (RRM) and mobility management. However, questions related to user-network association have not been explored much. Premkumar et al. in  propose a near optimal solution for a hybrid network within a combinatorial optimization framework, which is different from our approach. To the best of our knowledge, ours is the first attempt to present a generic formulation of the user-network association problem under a non-cooperative game framework. Moreover, this work is the first we know of that obtains an explicit threshold based policy for the WLAN-UMTS hybrid network model that we consider. II. F RAMEWORK FOR THE D ECISION C ONTROL P ROBLEM
The third generation wireless network deployed in this project is referred to UMTS system, which has been implemented through W-CDMA air interface. There are number of transition from 2G to 3G, starting from planning to field validation of the radio terminal. The crucial benefit of the 3Gnetwork is they are backward compactable with GSM network and this is achieved because the architecture of the radio terminal in both GSM and UMTS are designed and implemented through Software Defined Radio (SDR). Beyond the architecture and protocol stack of the wireless networks, the QoS (Quality of Service) depends on the cell dimensioning through the air interface, transmission dimensioning, coverage planning, capacity planning and traffic balancing. To maintaining the planned radio coverage optimal usage of the radio resource is very important to deliver the QoS . Today in mobile communication industries 80 percent of the total investment cost is spent for radio access network installation and for optimization to deliver QoS to customers.
Public communication networks such as UMTS, are basically configured as best effort, where all services share network resources equally; they get the same bit rate and experience the same delay. Network dimensioning is done in such a way that, bit rate or delay fulfils the most stringent requirements of the services provided in that network. Consequently, other services such as the background type of service, enjoys same quality, which is unnecessarily good and wastage of network resources . Moreover, traffic profiles across these networks are widely heterogeneous with respect to volumes of data per customer and QoS requirements. There is definitely unbalance between light users on voice only service or reading e-mail and heavy users on video streaming or multimedia-based video conferencing. When the network is loaded, this unbalanced usage of network resources can result into heavy users dominating the network. In such a situation, deploying e-learning services in a network configured on best effort basis, can compromise their performance due to presence of traffic from other users. Therefore, it is useful to take necessary measures to guarantee QoS for the mission-critic applications such as e-learning.
Much of interest has been involved in integrating UMTS and WLAN networks in order to take full advantage of the two. In such an integrated network, UMTS subscribers can access to UMTS services through 3G or WLAN access networks. This is highly desirable because network operators will use WLAN hotspots to increase network coverage, while offering subscribers with high-speed connections to 3G networks. To achieve this goal, a mechanism for service continuity between UMTS and WLAN is required. 3GPP has defined architecture for WLAN access to 3GPP system, but intersystem handover has not been yet investigated. Based on this architecture, this paper proposes a solution for seamless handover between UMTS and WLAN. Results obtained from the implementation of the proposed mechanism on our experimental system are also described.
A Rake receiver can decode several signals simultaneously and combine them to improve the quality of the signal or to get several services at the same time. In radio communications, the 10 strength of a signal can decrease for many reasons. Natural obstacles such as buildings and hills cause reflections, diffractions and scattering. Consequently, multipath propagation occurs which means that the same radio signal arrives at the receiver through different reflected paths. The Rake receiver uses the inherent frequency diversity characteristics of WCDMA as a means of providing redundancy in the network. Because the signal is spread over a wide frequency band, it is transmitted and received simultaneously on two or more frequencies. The Rake
There has been quite a lot of research done in the past, involv- ing the quality of VoIP in 3G system and evaluated the perfor- mance analysis and QoS. Several techniques that are be- lieved to bring improvements in VoIP quality have been pro- posed and techniques of how to control call and data conges- tion due to QoS factors. To initiate a VoIP call, at least, signal- ing protocols, that include, Session Initiation Protocol (SIP), H.323, H.248 (MEGACO) and MGCP  are required. SIP is defined in [4, 5]. In VoIP technology, a codec is essential for encoding and decoding speech. There are many types of co- dec's that can be used for this function.  proposed packet loss reduction to VoIP by means of AMR codec speech, whe- reby AMR codec maintains the toll quality of speech signals. According to , an AMR codec is a compulsory codec for conversational speech services within 3G systems. AMR co- dec consists of eight bit-rates which range from 4.5 kbps to 12.2 kbps and it is able to switch its bit-rate every 20 ms of speech frame depending on channel and network conditions .  proposed an end-to-end quality of service analysis in VoIP over 3G networks, whereby they checked if jitter in 3G networks have a negative effect on the end-user voice quality. According to , an E-model technique evaluates the quality of VoIP in wireless networks. This E-model technique accepts a wide range of telephone damages into consideration, like damages due to low-bit rate coding, one-way delay, echo and noise . In  QoS in VoIP over 3Gnetwork and Pricing Strategy , guaranteeing end-to-end quality to a VoIP call over UMTSnetwork is proposed, whereby the VoIP application pa- rameters (voice codec, packet size and de-jittering delay), and UMTS air interface parameters (coding rate
In summary, the network does not exhibit problems in behaviour until 12 packets (saturation factor 0.9) per second are offered by the application to the network; for 14 packets per second the sender still has no problems to send them, but the receiver is having some problems to receive them in a continuous way. This is seen by the fact that the average IPD is rising at the receiver, which means that there are less packets per second received. Furthermore, from saturation factor 1.0 on, the delay is increasing, packets are probably buffering somewhere before arriving at the receiver. In Figure 4.4 one can see that the sender can not get the expected number of packets send through in one second; the number of packets are lower at the receiver than at the sender, this is shown by the lower throughput, in the throughput plots, at the receiver side. And besides the throughput plots, this behaviour is also exhibited in the throughput difference histograms. For a saturation factor lower than 1.0 these histograms show a shared bottleneck, while for the saturation factors of 1.0 and higher, this shared bottleneck disappeared completely. For these reasons, we further focus on measurement data obtained for the saturation factor 0.9 only, corresponding to 12 packets per second; we continue with these measurements analysis to find answers to the research questions posed in this thesis.
Figure 6 presents the cdf (cumulative distribution function) of the positioning accuracy reported by the PCM. Assess- ment of the accuracy in the micro-/macrocellular urban and macrocellular suburban environments is executed by locat- ing the UE moving along two defined routes (indicated as solid and dashed lines in Figures 6(a) and 6(c)). Conducted measurements in the urban environment provide promising accuracy results (Figure 6(b)), since the accuracy for 67% of measurements is maintained below 70 m. At the same time, the reported 90% CERP is from 130 m in case of the route 1 to 180 m in the case of the route 2. The accuracy re- ported by the mobile travelling along the route 2 is evidently worse due to more locations close to the cell edge where the probability of erroneous estimation is higher, as pilots are hearable at similar levels in adjacent positioning regions. The achieved precision fulfils the defined FCC safety require- ments for network-based solutions with a big margin and si- multaneously it is su ﬃ cient for most of the location-sensitive applications. Similarly, in the case of the PCM operation in the typical macrocellular network, the accuracy is still main- tained at a good level. However, due to larger site spacing distances and definition of larger sizes of positioning regions, the error is higher compared to the reported accuracy in the dense urban network. As indicated in Figure 6(d), the accu- racy for 67% of measurements is reported at the level from 170 m to 190 m. Since the resolution of the PCM positioning in the considered macrocellular topology is limited by char- acterization of the positioning region size (100 m × 100 m), it is expected that for LBS requiring higher accuracy, the pre- cision of estimation could be further improved by adequate definition of positioning regions.
access network based on complex network are investi- gated. A cooperative caching framework named CNSC- RAN is proposed including MBS level, SBS level, and UE level. Based on CNSC-RAN, the functional modules including MBS controller, SBS controller, and UE con- troller are presented. Static and dynamic content- oriented network slicing procedure is illustrated. The process is designed to get the content required by UEs. In order to obtain the optimized resources sliced to each content in the framework, the content-oriented slicing is modeled and analyzed by using complex net- work and optimization theory and is formulated to minimize the average system cost to get the contents required by users in a known network architecture by using ER model and BA model. The problem is solved by a heuristic algorithm named CCSOA. The perform- ance of CCSOA is evaluated by the metrics including hit rate, average cache occupation, and average system cost as well as request traffic reduction to MBS in dy- namic content-oriented network slicing procedure en- abling UEs with self-evicting contents. As future work, we plan to investigate the performance of network sli- cing optimization on content caching considering the effect of user mobility.
While the design of GTP is lack of security concern, there are no embedded security schemes in GTP, so it has very obvious security vulnerabilities which can be easily exploited by attackers. With the widely usage of GTP, attacks toward this protocol could come from different directions, such as the air interface, internet and other PLMN (Public Land Mobile Network), so the attacks could make very huge damage not only to the core network infrastructure, but also to the internet and mobile users. According to this issue, in this paper, we briefly analyze the security issues of the GTP protocol in the first place, then, we propose a defense solution based on an event-based attack detection engine, and finally, the prototype system and experiment results are presented to show the effectiveness of such solution.
Future of wireless communication system designed to merge a variety of services such as voice, data, image and video. These services have various demands on the bandwidth and data rate. So studies has been done to access these requirements, Software Define Radio ( SDR) is being presented as technique offers the possible revolutions the way radios are manufacturing dissemination and used SDR promises to increase flexibility , expand hardware life time and lower costs . In this paper the simulation is being obtained by using MATLAB program for UMTS digital up-convertor to increase data rate as application of SDR.
In a mobile cellular network it is often necessary to transmit the same information to all the users (broadcast transmis- sion) or to a selected group of users (multicast transmission). Depending on the communication link conditions some re- ceivers will have better signal-to-noise ratios (SNR) than oth- ers and thus the capacity of the communication link for these users is higher. Cover  showed that in broadcast transmis- sions it is possible to exchange some of the capacity of the good communication links to the poor ones and the trade- o ﬀ can be worthwhile. A possible method to improve the ef- ficiency of the network is to use nonuniform signal constel- lations (also called hierarchical constellations) which are able to provide unequal bit error protection. In this type of con- stellations there are two or more classes of bits with diﬀer- ent error protection, to which diﬀerent streams of informa- tion can be mapped. Depending on the channel conditions, a given user can attempt to demodulate only the more pro- tected bits or also the other bits that carry the additional in- formation. An application of these techniques is in the trans- mission of coded voice or video signals. Several papers have studied the use of nonuniform constellations for this pur- pose [1, 2]. Nonuniform 16-QAM and 64-QAM constella- tions are already incorporated in the DVB-T (digital video broadcasting-terrestrial) standard .
In effect, these are two related but separate product sectors, each of which generate significant market volume. Market volume helps reduce component cost but also tends to improve component performance through better control of component toler- ances on the production line. Digital camera performance drives user expectation of how a digital cellular handset with an integrated digital camera will perform. The problem is that the digital cellular handset also has to be able to send and receive pic- tures and an audio stream over a radio physical layer that will typically consume sev- eral hundred milliWatts. There is a balance to be made between memory bandwidth in the handset and how much power to dedicate to sending and receiving image band- width, which in turn determines the user experience and user expectations.
In this paper, we use RTP/RTCP over unreliable UDP to realize the transportation of real time streaming. Meanwhile, according to the transmission feedback and network parameters, we analyze and calculate the network delay, packet loss and RTT (round trip time) to determine the network state. Finally, we propose a streaming media transmission con- trol scheme which could sense the network state and quickly adjust the rate of the sending side. It takes congestion, packet loss rate into comprehensive consideration and improves the overall performance of 3Gnetwork stream media. Keywords: RTP; Delay; RTT; Control Mechanism; 3G
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kinds. Cellular networks are typically characterized by limited ra- dio resources and significant device power consumption for net- work communications. The battery capacity of smartphones cannot be easily improved due to physical constraints in size and weight. Hence, battery life remains a key determinant of end-user expe- rience. Given the limited radio resources in these networks and device battery capacity constraints, optimizing the usage of these resources is critical for cellular carriers and application developers. In 3G and 4G cellular networks, the user equipment (UE) must stay in a high-power state, occupying radio resources for some re- quired time before the allocated resource is released by the net- work, and then the UE enters a low power state. This required time period, also known as the Radio Resource Control (RRC) tail , is necessary and important for cellular networks to prevent frequent state promotions (resource allocation), which can cause unaccept- ably long delays for the UE, as well as additional processing over- heads for the radio access network [4, 12]. Today’s cellular carriers use a static and conservative setting of the tail time in the order of many seconds, and previous studies have revealed this tail time to be the root cause of energy and radio resource inefficiencies in both 3G [15, 6, 11, 7] and 4G networks . Various optimization solu- tions have been proposed to address this problem, e.g., the use of fast dormancy [2, 3, 16] and client-side traffic shaping and schedul- ing [14, 18, 6]. In addition, specialized energy saving techniques for mobile applications have been proposed for specific applica- tions [21, 10] and for specific protocols .
Abstract— The performance of the Frequency Division Duplex (FDD) mode of the Code Division Multiple Access (CDMA) based Universal Mo- bile Telecommunication System (UMTS) is investigated. The new call blocking and call dropping probabilities, the probability of low-quality ac- cess as well as the required average transmit power are quantified both with and without the assistance of adaptive antenna arrays as well as with and without encountering shadow fading. In some of the scenarios investigated the system’s user capacity is doubled with the advent of adaptive antennas.
The crucial issues in cellular network for service provider is to provide the guaranteed quality of service (QoS), minimizing the dropping rate (DR) for handoff calls, blocking rate (BR) for new calls and most important is to increasing the capacity of network . This paper purposes two concepts in WCDMA network i.e call admission control (CAC) by considering distance of the cell as a deciding factor in order to minimize BR and DR .Our work is useful in the case when call can not transfer to lightly loaded cell as done in load balancing with cell breathing concept and blocked due to non availability of bandwidth then bandwidth degradation scheme (BDS) is used to increase the capacity of a cell/overall network and also to assign priority to the handoff calls over new call.
In this paper, we consider a cognitive radionetwork containing two cognitive radios (CRs) and one primary user. The CRs utilize finite number of received data samples for estimating the energy of the primary signals and forward these energy estimates to a fusion center (FC). The FC combines the energy estimates and utilizes a global threshold based on the exact knowledge of local thresholds of the CRs for determining the presence or absence of the primary signal. We propose selective and semi-selective soft combining schemes for this set-up. For the proposed schemes, we derive the total probability of error of detecting a spectrum hole. By minimizing the total probability of error in sensing a spectrum hole, we find optimized local and global thresholds. Moreover, we also discuss the optimization of conventional non-selective soft and 1-bit hard combining schemes with multiple (equal to or more than two) collaborative CRs under the total probability of error minimization criterion. It is shown by simulations that the proposed selective soft combination-based scheme significantly outperforms the conventional non-selective schemes based on soft combination and 1-bit hard combination. Further, it is shown by simulation that the proposed selective soft combining scheme along with the total probability of error minimization criterion is able to properly utilize a spectrum hole with interference level less than the standard specified value.