Network Simulator (network + simulator)

Distribution by Scientific Domains


Selected Abstracts


Adaptive measurement-based traffic engineering in small differentiated services domains

EUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 1 2007
Sven Krasser
In this paper, we propose a framework for measurement-based traffic engineering and connection admission control in small-differentiated services domains. The domain investigated is a wired radio access network based on the Internet protocol (IP). This framework is evaluated by simulation using the popular network simulator ns-2. The framework is adaptive to changes in the network load and supports multiple types of service. All traffic-engineering decisions are made by edge routers (ERs) at the rim of the network domain. Multiple disjoint paths are configured between those ERs. Network state information is gathered in two different fashions. We evaluate a scheme based on the states of the queues on each alternative path and a scheme based on end-to-end probe packet transmission characteristics on each alternative path. Both schemes are compared to a shortest path first (SPF) routing approach. Copyright © 2006 AEIT [source]


TCP-friendly transmission of voice over IP

EUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 3 2003
F. Beritelli
In the last few years an increasing amount of attention has been paid to technologies for the transmission of voice over IP (VoIP). At present, the UDP transport protocol is used to provide this service. However, when the same bottleneck link is shared with TCP flows, and in the presence of a high network load and congestion, UDP sources capture most of the bandwidth, strongly penalizing TCP sources. To solve this problem some congestion control should be introduced for UDP traffic as well, in such a way that this traffic becomes TCP-friendly. In this perspective, several TCP-friendly algorithms have been proposed in the literature. Among them, the most promising candidates for the immediate future are RAP and TFRC. However, although these algorithms were introduced to support real-time applications on the Internet, up to now the only target in optimizing them has been that of achieving fairness with TCP flows in the network. No attention has been paid to the applications using them, and in particular, to the quality of service (QoS) perceived by their users. The target of this paper is to analyze the problem of transmitting voice over IP when voice sources use one of these TCP-friendly algorithms. With this aim, a VoIP system architecture is introduced and the characteristics of each its elements are discussed. To optimize the system, a multirate voice encoder is used so as to be feasible to work over a TCP layer, and a modification of both RAP and TFRC is proposed. Finally, in order to analyze the performance of the proposed system architecture and to compare the modified RAP and TFRC with the original algorithms, the sources have been modeled with an arrival process modulated by a Markov chain, and the model has been used to generate traffic in a simulation study performed with the ns-2 network simulator. Copyright © 2003 AEI. [source]


An opportunistic cross-layer architecture for voice in multi-hop wireless LANs

INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 4 2009
Suhaib A. Obeidat
Abstract We propose an opportunistic cross-layer architecture for adaptive support of Voice over IP in multi-hop wireless LANs. As opposed to providing high call quality, we target emergencies where it is important to communicate, even if at low quality, no matter the harshness of the network conditions. With the importance of delay on voice quality in mind, we select adaptation parameters that control the ratio of real-time traffic load to available bandwidth. This is achieved in two ways: minimizing the load and maximizing the bandwidth. The PHY/MAC interaction improves the use of the spectral resources by opportunistically exploiting rate-control and packet bursts, while the MAC/application interaction controls the demand per source through voice compression. The objective is to maximize the number of calls admitted that satisfy the end-to-end delay budget. The performance of the protocol is studied extensively in the ns-2 network simulator. Results indicate that call quality degrades as load increases and overlonger paths, and a larger packet size improves performance. For long paths having low-quality channels, forward error correction, header compression, and relaxing the delay budget of the system are required to maintain call admission and quality. The proposed adaptive protocol achieves high performance improvements over the traditional, non-adaptive approach. Copyright © 2008 John Wiley & Sons, Ltd. [source]


Linear discriminant analysis in network traffic modelling

INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 1 2006
Bing-Yi Zhang
Abstract It is difficult to give an accurate judgement of whether the traffic model fit the actual traffic. The traditional method is to compare the Hurst parameter, data histogram and autocorrelation function. The method of comparing Hurst parameter cannot give exact results and judgement. The method of comparing data histogram and autocorrelation only gives a qualitative judgement. Based on linear discriminant analysis we proposed a novel arithmetic. Utilizing this arithmetic we analysed some sets of data with large and little differences. We also analysed some sets of data generated by network simulator. The analysis result is accurate. Comparing with traditional method, this arithmetic is useful and can conveniently give an accurate judgement for complex network traffic trace. Copyright © 2005 John Wiley & Sons, Ltd. [source]


A batch-type time-true ATM-network simulator,design for parallel processing

INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 8 2002
Michael Logothetis
Abstract This paper presents a new type of network simulator for simulating the call-level operations of telecom networks and especially ATM networks. The simulator is a pure time-true type as opposed to a call-by-call type simulator. It is also characterized as a batch-type simulator. The entire simulation duration is divided into short time intervals of equal duration, t. During t, a batch processing of call origination or termination events is executed and the time-points of these events are sorted. The number of sorting executions is drastically reduced compared to a call-by-call simulator, resulting in considerable timesaving. The proposed data structures of the simulator can be implemented by a general-purpose programming language and are well fitted to parallel processing techniques for implementation on parallel computers, for further savings of execution time. We have first implemented the simulator in a sequential computer and then we have applied parallelization techniques to achieve its implementation on a parallel computer. In order to simplify the parallelization procedure, we dissociate the core simulation from the built-in call-level functions (e.g. bandwidth control or dynamic routing) of the network. The key point for a parallel implementation is to organize data by virtual paths (VPs) and distribute them among processors, which all execute the same set of instructions on this data. The performance of the proposed batch-type, time-true, ATM-network simulator is compared with that of a call-by-call simulator to reveal its superiority in terms of sequential execution time (when both simulators run on conventional computers). Finally, a measure of the accuracy of the simulation results is given. Copyright © 2002 John Wiley & Sons, Ltd. [source]


Performance evaluation of LIBTA/hybrid time-slot selection algorithm for cellular systems,

INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 6 2001
Jyh-Horng Wen
Abstract This paper studies the performance of radio assignment algorithms for portable access in cellular systems. Several channel access procedures are proposed and simulated using block oriented network simulator (BONeS) simulation of a model 36-port system. Simulation results exhibit that load-sharing system with LIBTA algorithm is better than directed retry system with the same algorithm by around 0.9 erlangs while better than quasi-fixed channel assignment (QFCA) system by around 2 erlangs if the grade of service (GOS) is constrained to less than 10 per cent. Plus, a hybrid time-slot selection procedure is proposed to enhance the system performance. It is observed that systems with hybrid time-slot selection perform better than those with LIBTA algorithm in GOS under heavy load. It is also observed that load sharing system with hybrid time-slot selection algorithm is better than directed retry system with the same algorithm by around 0.7 erlangs and better than QFCA system by around 2 erlangs. Copyright © 2001 John Wiley & Sons, Ltd. [source]


An OPNET-based simulation approach for deploying VoIP

INTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 3 2006
K. Salah
These days a massive deployment of VoIP is taking place over IP networks. VoIP deployment is a challenging task for network researchers and engineers. This paper presents a detailed simulation approach for deploying VoIP successfully. The simulation uses the OPNET network simulator. Recently OPNET has gained a considerable popularity in both academia and industry, but there is no formal or known approach or methodology as to how OPNET can be used to assess the support and readiness of an existing network in deploying VoIP. Our approach and work presented in this paper predict, prior to the purchase and deployment of VoIP equipment, the number of VoIP calls that can be sustained by an existing network while satisfying QoS requirements of all network services and leaving adequate capacity for future growth. As a case study, we apply the simulation approach on a typical network of a small enterprise. The paper presents a detailed description of simulation models for network topology and elements using OPNET. The paper describes modeling and representation of background and VoIP traffic, as well as various simulation configurations. Moreover, the paper discusses many design and engineering issues pertaining to the deployment of VoIP. These issues include characteristics of VoIP traffic and QoS requirements, VoIP flow and call distribution, defining future growth capacity, and measurement and impact of background traffic.,Copyright © 2006 John Wiley & Sons, Ltd. [source]


Integrating the scene length characteristics of MPEG video bitstreams into a direct broadcast satellite network with return channel system

INTERNATIONAL JOURNAL OF SATELLITE COMMUNICATIONS AND NETWORKING, Issue 2 2004
Fatih Alagöz
Abstract In order to optimize the network resources, we should incorporate all the available information into the network design. However, incorporating irrelevant information may increase the design complexity and/or decrease the performance of the network. In this paper, we investigate the relevance of integrating the scene length characteristics of moving pictures expert group (MPEG) coded video bitstreams into a direct broadcast satellite (DBS) network with return channel system (DVB-RCS). Due to the complexity of the studied system, unless disputable simplifications are made, it is hard to achieve a mathematical foundation for this integration. Our analysis relies on extensive set of simulations. Firstly, we achieve the scene length distributions for MPEG bitstreams based on the proposed scene change models and their subjective observations of the actual video. We show that these models may be used to estimate the scene length of MPEG bitstreams. We then integrate this estimation into a DBS network simulator. Finally, we show that the scene length characteristics may be used to improve the DBS network performance under certain conditions. Copyright © 2004 John Wiley & Sons, Ltd. [source]