A unified performance model for reservation-type multiple-access schemes

This paper presents a unified performance model for the integrated voice/data wireless system using reservationtype multiple-access (MA) schemes. It is observed that although these schemes are different in the frame structure and resource assignment procedure, all of them can be described by several common state variables whose evolvement exhibits the Markovian property. Based on this observation, a general Markovian model is developed in this paper. Three performance measures, namely, voice-packet-loss probability, data throughput, and data delay are defined. As a special case, the performance evaluation model for the voice-only system is also presented. Numerical results are given and verified by simulation under both voice-only and integrated scenarios using packet-reservation MA (PRMA), dynamic time-division multiple access (D-TDMA), and resourceauction multiple access (RAMA) as examples. It is found that our analytical model is quite accurate, especially in the region of interest. The impact of system parameters (such as the voice-permission probability, data-retransmission probability, maximum number of voice slots per frame, etc.) on the integrated system performance is also investigated for these three example systems. © 1998 IEEE.published_or_final_versio

Packet Scheduling Study for Heterogeneous Traffic in Downlink 3GPP LTE System

Long Term Evolution (LTE) network deploys Orthogonal Frequency Division Multiple Access (OFDMA) technology for downlink multi-carrier transmission. To meet the Quality of Service (QoS) requirements for LTE networks, packet scheduling has been employed. Packet scheduling determines when and how the user’s packets are transmitted to the receiver. Therefore effective design of packet scheduling algorithm is an important discussion. The aims of packet scheduling are maximizing system throughput, guaranteeing fairness among users, andminimizing either or both PacketLoss Ratio (PLR)and packet delay. Inthis paper, the performance of two packet scheduling algorithms namely Log Maximum-Largest Weighted Delay First (LOG-MLWDF) and Max Delay Unit (MDU), developed for OFDM(Orthogonal Frequency Division Multiplexing)networks, has been investigated in LTE downlink networks, and acomparison of those algorithmswith a well-known scheduling algorithm namely Maximum-Largest Weighted Delay First(MLWDF) has been studied.The performance evaluation was in terms of system throughput, PLR and fairness index. This study was performed forboth real time (voice and video streaming)and non-real time (best effort)perspectives. Results show that for streaming flows,LOG-MLWDF shows best PLR performance among the considered scheduling schemes, and for best effort flows, it outperforms theother two algorithms in terms of packet delay and throughput

Security of Streaming Media Communications with Logistic Map and Self-Adaptive Detection-Based Steganography

Voice over IP (VoIP) is finding its way into several applications, but its security concerns still remain. This paper shows how a new self-adaptive steganographic method can ensure the security of covert VoIP communications over the Internet. In this study an Active Voice Period Detection algorithm is devised for PCM codec to detect whether a VoIP packet carries active or inactive voice data, and the data embedding location in a VoIP stream is chosen randomly according to random sequences generated from a logistic chaotic map. The initial parameters of the chaotic map and the selection of where to embed the message are negotiated between the communicating parties. Steganography experiments on active and inactive voice periods were carried out using a VoIP communications system. Performance evaluation and security analysis indicates that the proposed VoIP steganographic scheme can withstand statistical detection, and achieve secure real-time covert communications with high speech quality and negligible signal distortion

Design and performance analysis of an Integrated Voice/Data (IVD) protocol for a token ring network

A high performance integrated voice/data (IVD) protocol for token ring networks that operates over a wide range of data traffic levels is developed and analyzed in this dissertation;The problems raised by integrating voice/data in local area networks are identified. These problems include variable network access delay and end-to-end delay limit of voice packets, and queueing delay of data packets;In the design of the IVD protocol, a packet format is selected, conditions for selecting network parameters are derived, and a channel allocation strategy is described to provide high quality of voice over a wide range of data traffic levels while preserving satisfactory data performance;The implementation issues, specification, and operation of the proposed protocol are described based on the standard IEEE 802.5 token ring protocol. The implementation issues are studied for two principal purposes: not to modify the operating data protocol and to utilize the bandwidth for data packets when voice stations are idle. A state transition diagram is used to specify the proposed protocol;A discrete-event model of the proposed IVD protocol is developed for the accurate performance evaluation of data and real-time voice traffic. In modeling the protocol, all the necessary information which affects the state of the system are considered including location of stations;The voice/data performance of the proposed IVD token ring protocol is evaluated and the effects of voice on data and vice versa are studied. The proposed protocol provides high quality of voice without a large degradation on the data performance over a wide range of data loads

Performance evaluation of TCP, UDP and DCCP for video traffics over 4G network

Fourth Generation (4G) system has been used more widely than the older generations 3G and 2G. Among the reasons are that the 4G’s transfer rate is higher and it supports all multimedia functions. Besides, its’ supports for wide geographical locus makes wireless technology gets more advanced. The essential goal of 4G is to enable voice-based communication being implemented endlessly. To achieve the goal, this study tries to answer the following research questions: (1), are the old protocols suit with this new technology; (2), which one has the best performance and, (3) which one has the greatest effect on throughput, delay, packet delivery ratio and packet loss. The aforementioned questions are crucial in the performance evaluation of the most famous protocols (particularly User Datagram Protocol (UDP), Transmission Control Protocol (TCP), and Datagram Congestion Control Protocol (DCCP)) within the 4G environment. Through the Network Simulator-3 (NS-3), the performance of transporting MPEG-4 video stream including throughput, delay, packet loss, and packet delivery ratio are analyzed at the base station through UDP, TCP, and DCCP protocols over 4G’s Long Term Evolution (LTE) technology. The results show that DCCP has better throughput, and lesser delay, but at the same time it has more packet loss than UDP and TCP. Based on the results, DCCP is recommended as a transport protocol for real time vide

Approximate performance analysis of slotted downlink channel in a wireless CDMA system supporting integrated voice and data services, Journal of Telecommunications and Information Technology, 2004, nr 2

This paper is concerned with the performance analysis of a slotted downlink channel in a wireless CDMA communication system with integrated packet voice and data transmission. The system model consists of mobile terminals (MT) and a single base station (BS). It is assumed that the voice (data) packet error rate (PER) does not exceed 10-2 (10-5).With this requirement the number of simultaneous transmissions over the downlink channel is limited. Therefore, the objective of the call admission control is to restrict the maximum number of CDMA codes available to voice and data traffic. Packets of accepted voice calls are transmitted immediately while accepted data packets are initially buffered at the BS. This station distinguishes between silence and talk-spurt periods of voice sources, so that data packets can use their own codes for transmission during silent time slots. Data packets are buffered in queues created separately for each destination. Discrete-time Markov processes are used to model the system operation. Statistical dependence between queues is the main difficulty which arises during the analysis. This dependence leads to serious computational complexity. The aim of this paper is to present an approximate analytical method based on the restricted occupancy urn model which enables to evaluate system performance despite the dependence. Numerical calculations compared with simulation results show excellent agreement for the average system throughput and the blocking probability of data packets for higher system loads. On the other hand, when the average data packet delay is considered, analytical results underestimate simulation and therefore only approximate system performance evaluation is possible

Performance evaluation of voice handover between LTE and UMTS

M.Sc.(Eng.), Faculty of Engineering and the Built Environment, 2011The main objective of seamless mobility is to enable mobile users to stay connected while roaming across heterogeneous networks. As cellular networks evolve from the third generation Universal Mobile Telecommunication System (UMTS) to the Long Term Evolution (LTE), a new Evolved Packet Core (EPC) will support heterogeneous radio access networks on the same platform. UMTS provides voice services in the circuit switched domain; while LTE operates in the packet switched domain. Cellular network operators thus face the challenge of providing voice services during initial deployment of LTE due to difficulty in mobility between the two domains. Seamless voice handover between packet switched LTE and the circuit switched UMTS network is therefore an important tool in solving this problem. This report investigates the performance of inter-Radio Access Technology voice handover between LTE and UMTS. The schemes evaluated were Voice Call Continuity (VCC) for UMTS to LTE handover and Single Radio Voice Call Continuity (SRVCC) for LTE to UMTS handover. The performance evaluation was done using mathematical models and equations that were derived for the handover service interruption time. The resulting equations were simulated and the output was analysed and compared with the Third Generation Partnership Project (3GPP) specifications

Final report on the evaluation of RRM/CRRM algorithms

Deliverable public del projecte EVERESTThis deliverable provides a definition and a complete evaluation of the RRM/CRRM algorithms selected in D11 and D15, and evolved and refined on an iterative process. The evaluation will be carried out by means of simulations using the simulators provided at D07, and D14.Preprin

Exponential MLWDF (EXP-MLWDF) Downlink Scheduling Algorithm Evaluated in LTE for High Mobility and Dense Area Scenario

Nowadays, with the advent of smartphones, most of people started to make voice and video conference calls continuously even in a high mobility scenario, the bandwidth requirements have increased considerably, which can cause network congestion phenomena. To avoid network congestion problems and to support high mobility scenario, 3GPP has developed a new cellular standard based packet switching, termed LTE (Long Term Evolution). The purpose of this paper is to evaluate the performance of the new proposed algorithm, named Exponential Modified Largest Weighted Delay First ‘EXP-MLWDF’, for high mobility scenario and with the presence of a large number of active users, in comparison with the well-known algorithms such as a proportional fair algorithm (PF), Exponential Proportional Fairness (EXP/PF), Logarithm Rule (LOG-Rule), Exponential Rule (EXP-Rule) and Modified Largest Weighted Delay First (MLWDF). The performance evaluation is conducted in terms of system throughput, delay and PLR. Finally, it will be concluded that the proposed scheduler satisfies the quality of service (QoS) requirements of the real-time traffic in terms of packet loss ratio (PLR), average throughput and packet delay. Because of the traffic evolution, some key issues related to scheduling strategies that will be considered in the future requirements are discussed in this article