776 research outputs found
Real-time characteristics of switched ethernet for "1553B" -embedded applications : simulation and analysis
In our previous work , Full Duplex Switched Ethernet was put forward as an attractive candidate to replace the MIL-STD 1553B data bus, in next generation "1553B"-embedded applications. An analytic study was conducted, using the Network Calculus formalism, to evaluate the deterministic guarantees offered by our proposal. Obtained results showed the effectiveness of traffic shaping techniques, combined with priority handling mechanisms on Full Duplex Switched Ethernet in order to satisfy 1553B-like real-time constraints. In this paper, we extend this work by the use of simulation. This gives the possibility to capture additional characteristics of the proposed architecture with respect to the analytical study, which was basically used to evaluate worst cases and deterministic guarantees. Hence, to assess the real-time characteristics of our proposed interconnection technology, the results yielded by simulation are discussed and average latencies distributions are considered
A Unified Mobility Management Architecture for Interworked Heterogeneous Mobile Networks
The buzzword of this decade has been convergence: the convergence of telecommunications, Internet, entertainment, and information technologies for the seamless provisioning of multimedia services across different network types. Thus the future Next Generation Mobile Network (NGMN) can be envisioned as a group of co-existing heterogeneous mobile data networking technologies sharing a common Internet Protocol (IP) based backbone. In such all-IP based heterogeneous networking environments, ongoing sessions from roaming users are subjected to frequent vertical handoffs across network boundaries. Therefore, ensuring uninterrupted service continuity during session handoffs requires successful mobility and session management mechanisms to be implemented in these participating access networks. Therefore, it is essential for a common interworking framework to be in place for ensuring seamless service continuity over dissimilar networks to enable a potential user to freely roam from one network to another. For the best of our knowledge, the need for a suitable unified mobility and session management framework for the NGMN has not been successfully addressed as yet. This can be seen as the primary motivation of this research. Therefore, the key objectives of this thesis can be stated as: To propose a mobility-aware novel architecture for interworking between heterogeneous mobile data networks To propose a framework for facilitating unified real-time session management (inclusive of session establishment and seamless session handoff) across these different networks. In order to achieve the above goals, an interworking architecture is designed by incorporating the IP Multimedia Subsystem (IMS) as the coupling mediator between dissipate mobile data networking technologies. Subsequently, two different mobility management frameworks are proposed and implemented over the initial interworking architectural design. The first mobility management framework is fully handled by the IMS at the Application Layer. This framework is primarily dependant on the IMS’s default session management protocol, which is the Session Initiation Protocol (SIP). The second framework is a combined method based on SIP and the Mobile IP (MIP) protocols, which is essentially operated at the Network Layer. An analytical model is derived for evaluating the proposed scheme for analyzing the network Quality of Service (QoS) metrics and measures involved in session mobility management for the proposed mobility management frameworks. More precisely, these analyzed QoS metrics include vertical handoff delay, transient packet loss, jitter, and signaling overhead/cost. The results of the QoS analysis indicates that a MIP-SIP based mobility management framework performs better than its predecessor, the Pure-SIP based mobility management method. Also, the analysis results indicate that the QoS performances for the investigated parameters are within acceptable levels for real-time VoIP conversations. An OPNET based simulation platform is also used for modeling the proposed mobility management frameworks. All simulated scenarios prove to be capable of performing successful VoIP session handoffs between dissimilar networks whilst maintaining acceptable QoS levels. Lastly, based on the findings, the contributions made by this thesis can be summarized as: The development of a novel framework for interworked heterogeneous mobile data networks in a NGMN environment. The final design conveniently enables 3G cellular technologies (such as the Universal Mobile Telecommunications Systems (UMTS) or Code Division Multiple Access 2000 (CDMA2000) type systems), Wireless Local Area Networking (WLAN) technologies, and Wireless Metropolitan Area Networking (WMAN) technologies (e.g., Broadband Wireless Access (BWA) systems such as WiMAX) to interwork under a common signaling platform. The introduction of a novel unified/centralized mobility and session management platform by exploiting the IMS as a universal coupling mediator for real-time session negotiation and management. This enables a roaming user to seamlessly handoff sessions between different heterogeneous networks. As secondary outcomes of this thesis, an analytical framework and an OPNET simulation framework are developed for analyzing vertical handoff performance. This OPNET simulation platform is suitable for commercial use
Xcast Based Routing Protocol For Push To Talk Application In Mobile Ad Hoc Networks
Mobile ad-hoc networks comprise a type of wireless network that can be easily
created without the need for network infrastructure or administration. These
networks are organized and administered into temporary and dynamic network
topologies. Unfortunately, mobile ad-hoc networks suffer from some limitations
related to insufficient bandwidth. The proliferation of new IP Multimedia subsystem
services (IMs), such as Push-to-talk (PTT) applications consume large amounts of
bandwidth, resulting in degraded QoS performance of mobile ad-hoc networks. In
this thesis, a Priority XCAST based routing protocol (P-XCAST) is proposed for
mobile ad-hoc networks to minimize bandwidth consumption. P-XCAST is based on
demand route requests and route reply mechanisms for every destination in the PXCAST
layer. To build the network topology and fill up the route table for nodes,
the information in the route table is used to classify the XCAST list of destinations
according to similarities on their next hop. Furthermore, P-XCAST is merged with a
proposed Group Management algorithm to handle node mobility by classifying nodes
into two types: group head and member. The proposed protocol was tested using the
GloMoSim network simulator under different network scenarios to investigate
Quality of Service (QoS) performance network metrics. P-XCAST performance was
better by about 20% than those of other tested routing protocols by supporting of
group size up to twenty receivers with an acceptable QoS. Therefore, it can be
applied under different network scenarios (static or dynamic). In addition Link
throughput and average delay was calculated using queuing network model; as this
model is suitable for evaluating the IEEE 802.11 MAC that is used for push to talk applications. The analytical results for link throughput and average delay were used
to validate the simulated results
Modelling the probability density function of IPTV traffic packet delay variation
This article deals with modelling the Probability density function of IPTV traffic packet delay variation. The use of this modelling is in an efficient de-jitter buffer estimation. When an IP packet travels across a network, it experiences delay and its variation. This variation is caused by routing, queueing systems and other influences like the processing delay of the network nodes. When we try to separate these at least three types of delay variation, we need a way to measure these types separately. This work is aimed to the delay variation caused by queueing systems which has the main implications to the form of the Probability density function
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Application-aware scheduling for VoIP in Wireless Mesh Networks
Wireless Mesh Networks (WMNs) are seen as a means to provide last mile connections in Next Generation Networks (NGNs). Because of their auto configuration capabilities and the low deployment cost WMNs are considered to be an efficient solution for the support of multiple voice, video and data services in NGNs. This paper looks at the optimal provision of resources in WMNs for Voice over IP (VoIP) traffic, which has strict performance requirements in terms of delay, jitter and packet loss. In WMNs, because of the challenges introduced by wireless multi hop transmissions and limited resources, providing performance quality for VoIP comparable to the voice quality in the traditional circuit switched networks is a major challenge.
This paper analyses different scheduling mechanisms for TDMA based access control in mesh networks as specified in the IEEE 802.16 2004 WiMAX standard. The performance of the VoIP applications when different scheduling mechanisms are deployed is analysed on a variety of topologies using ns 2 simulation and mathematical analysis. The paper concludes that on demand scheduling of VoIP traffic typically deployed in 802.11 based WMNs is not able to provide the required VoIP quality in realistic mesh WiMAX network scenarios and is therefore not optimal from a network operator’s point of view. Instead, it is shown, that continuous scheduling is much better suited to serve VoIP traffic. The paper then proposes a new VoIP aware resource coordination scheme and shows, through simulation, that the new scheme is scalable and provides good quality for VoIP service in a wide range of network scenarios. The results shown in the paper prove that the new scheme is resilient to increasing hop count, increasing number of simultaneous VoIP sessions and the background traffic load in the network. Compared to other resource coordination schemes the VoIP aware scheduler significantly increases the number of supported calls
Mobile Networks
The growth in the use of mobile networks has come mainly with the third generation systems and voice traffic. With the current third generation and the arrival of the 4G, the number of mobile users in the world will exceed the number of landlines users. Audio and video streaming have had a significant increase, parallel to the requirements of bandwidth and quality of service demanded by those applications. Mobile networks require that the applications and protocols that have worked successfully in fixed networks can be used with the same level of quality in mobile scenarios. Until the third generation of mobile networks, the need to ensure reliable handovers was still an important issue. On the eve of a new generation of access networks (4G) and increased connectivity between networks of different characteristics commonly called hybrid (satellite, ad-hoc, sensors, wired, WIMAX, LAN, etc.), it is necessary to transfer mechanisms of mobility to future generations of networks. In order to achieve this, it is essential to carry out a comprehensive evaluation of the performance of current protocols and the diverse topologies to suit the new mobility conditions
Process and tool support for real-time performance analysis of integrated modular systems
This paper describes a real-time system performance analysis methodology and toolset that has been developed at SEIC to be an
integral part of a broader BAE Systems Military Air Solutions (MAS) process and toolset for Integrated Modular Systems (IMS).
The proposed modelling approach and toolset components provide some key ‘through-life’ real-time system engineering benefits
relating to system performance, including : the ability to construct a performance prediction model during the early stages of
system design and to independently model the timing behaviour of end-to-end transactions across a distributed system of shared
processing and network resources
A novel non-intrusive objective method to predict voice quality of service in LTE networks.
This research aimed to introduce a novel approach for non-intrusive objective
measurement of voice Quality of Service (QoS) in LTE networks. While achieving this aim, the thesis established a thorough knowledge of how voice traffic is
handled in LTE networks, the LTE network architecture and its similarities and
differences to its predecessors and traditional ground IP networks and most
importantly those QoS affecting parameters which are exclusive to LTE environments. Mean Opinion Score (MOS) is the scoring system used to measure
the QoS of voice traffic which can be measured subjectively (as originally intended). Subjective QoS measurement methods are costly and time-consuming,
therefore, objective methods such as Perceptual Evaluation of Speech Quality
(PESQ) were developed to address these limitations. These objective methods
have a high correlation with subjective MOS scores. However, they either require individual calculation of many network parameters or have an intrusive
nature that requires access to both the reference signal and the degraded signal
for comparison by software. Therefore, the current objective methods are not
suitable for application in real-time measurement and prediction scenarios.
A major contribution of the research was identifying LTE-specific QoS affecting parameters. There is no previous work that combines these parameters to
assess their impacts on QoS.
The experiment was configured in a hardware in the loop environment. This
configuration could serve as a platform for future research which requires simulation of voice traffic in LTE environments.
The key contribution of this research is a novel non-intrusive objective method
for QoS measurement and prediction using neural networks. A comparative
analysis is presented that examines the performance of four neural network
algorithms for non-intrusive measurement and prediction of voice quality over
LTE networks. In conclusion, the Bayesian Regularization algorithm with 4 neurons in the hidden layer and sigmoid symmetric transfer function was identified as the best solution with a Mean Square Error (MSE) rate of 0.001 and
regression value of 0.998 measured for the testing data set
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