Block Reservation Time Division Multiple Access (BRTDMA) Protocol for a High Capacity Wireless Network

Next generation high capacity wireless networks need to support various types of traffic, including voice, video and data, each of which have different Quality of Service (QoS) requirements for successful transmission. This paper presents an advanced reservation packet access protocol BRTDMA (Block Reservation Time Division Multiple Access) that can accommodate voice and data traffic with equal efficiency in a wireless network. The proposed BRTDMA protocol has been designed to operate in a dynamic fashion by allocating resources according to the QoS criteria of voice and data traffic. Most of the existing reservation protocols offers reservation to voice traffic while data packets are transmitted using contention mode. In this paper we propose a block reservation technique to reserve transmission slots for data traffic for a short duration, which minimizes the speech packet loss and reduce the end-to-end delay for wireless data traffic. The optimum block reservation length for data traffic has been studied in a cellular mobile radio environment using a simulation model. Simulation results show that the BRTDMA protocol offers higher traffic capacity than standard PRMA protocol for integrated voice and data traffic and offers flexibility in accommodating multimedia traffic.     

Performance evaluation of the RQMA-CDMA scheme for multimedia traffic with QoS guarantees

The rapid growth of cellular mobile technology in recent years, coupled with the explosive growth of the Internet, has significantly increased the demand for wireless data services. Traffic on mobile devices is expected to be a mix of real-time traffic such as video, voice, and data, with users requiring diverse quality of service (QoS) guarantees for different types of traffic (video, voice and data). One of the primary challenges of providing QoS is how to prioritize and allocate network resources among contending applications. In order to achieve these goals, a scheduling scheme that can provide equitable and effective packet routing is required. This paper proposes a scheduling scheme called remote queuing multiple access-code division multiple access (RQMA-CDMA), whose purpose is to equitably assign bandwidth resources with QoS guarantees to different mobile devices. RQMA-CDMA is a rate scheduling scheme that can be used to assign bandwidth resources in conjunction with GPS (generalized processor sharing). Additionally, we analyze an admission control that is based on signal to interference plus noise ratio (SINR) for multimedia traffic. Finally, the performance of RQMA-CDMA is evaluated and compared to schemes based on CDMA-GPS according to dropped packets, delay, and throughput.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Novel Radio Link Buffer Management Schemes for End-User Multi-class Traffic in High Speed Packet Access Networks

The requirement to provide multimedia services with QoS support in mobile networks has led to standardization and deployment of high speed data access technologies such as the High Speed Downlink Packet Access (HSDPA) system. HSDPA improves downlink packet data and multimedia services support in WCDMA-based cellular networks. As is the trend in emerging wireless access technologies, HSDPA supports end-user multi-class sessions comprising parallel flows with diverse Quality of Service (QoS) requirements, such as real-time (RT) voice or video streaming concurrent with non real-time (NRT) data service being transmitted to the same user, with differentiated queuing at the radio link interface. Hence, in this paper we present and evaluate novel radio link buffer management schemes for QoS control of multimedia traffic comprising concurrent RT and NRT flows in the same HSDPA end-user session. The new buffer management schemes—Enhanced Time Space Priority (E-TSP) and Dynamic Time Space Priority (D-TSP)—are designed to improve radio link and network resource utilization as well as optimize end-to-end QoS performance of both RT and NRT flows in the end-user session. Both schemes are based on a Time-Space Priority (TSP) queuing system, which provides joint delay and loss differentiation between the flows by queuing (partially) loss tolerant RT flow packets for higher transmission priority but with restricted access to the buffer space, whilst allowing unlimited access to the buffer space for delay-tolerant NRT flow but with queuing for lower transmission priority. Experiments by means of extensive system-level HSDPA simulations demonstrates that with the proposed TSP-based radio link buffer management schemes, significant end-to-end QoS performance gains accrue to end-user traffic with simultaneous RT and NRT flows, in addition to improved resource utilization in the radio access network.     

Performance evaluation of a CDMA protocol for voice and dataintegration in personal communication networks

Future wireless personal communication networks (PCN's) will require voice and data service integration on the radio link. The multiaccess capability of the code-division multiple-access (CDMA) technique has been widely investigated in the recent literature. The aim of this paper is to propose a CDMA-based protocol for joint voice and data transmissions in PCN's. The performance of such a protocol has been derived by means of an analytical approach both in terms of voice packet dropping probability and mean data packet delay. Voice traffic has been modeled as having alternated talkspurts and silences, with generation of voice packets at constant rate during talkspurts and no packet generation during silence gaps. A general arrival process is assumed for the data traffic. However, numerical results are derived in the case of a Poisson process. Simulation results are given to validate our analytical predictions. The main result derived here is that the proposed CDMA-based protocol efficiently handles both voice and data traffic. In particular, it is shown that the performance of the voice subsystem is independent of the data traffic     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Secured Self Organizing Network Architecture in Wireless Personal Networks

Secured self organizing network is an approach to computer network architecture that seeks to address the technical issues in heterogeneous networks that may lack continuous network connectivity. In delay tolerant network packets storage exists when there is any link breakage between the nodes in the network so delay is tolerable in this type of network during the data transmission. But this delay is not tolerable in wireless network for voice packet transmission. This evokes the use of wireless networks. In a network, different wireless network topologies are interoperating with each other so the communication across the network is called overlay network. This network is vulnerable to attacks due to mobile behaviour of nodes and frequent changes in topologies of the network. The attacks are wormhole attack and blackhole attack is analysed in this paper. They are critical threats to normal operation in wireless networks which results in the degradation of the network performance. The proposed recovery algorithm for wormhole and the isolation of blackhole will increase the performance of the network. The performance metrics such as throughput, packet delivery ratio, end–end delay and routing overhead of the network are evaluated.

     

Improving Voice‐Service Support in Cognitive Radio Networks

Voice service is very demanding in cognitive radio networks (CRNs). The available spectrum in a CRN for CR users varies owing to the presence of licensed users. On the other hand, voice packets are delay sensitive and can tolerate a limited amount of delay. This makes the support of voice traffic in a CRN a complicated task that can be achieved by devising necessary considerations regarding the various network functionalities. In this paper, the support of secondary voice users in a CRN is investigated. First, a novel packet scheduling scheme that can provide the required quality of service (QoS) to voice users is proposed. The proposed scheme utilizes the maximum packet transmission rate for secondary voice users by assigning each secondary user the channel with the best level of quality. Furthermore, an analytical framework developed for a performance analysis of the system, is described in which the effect of erroneous spectrum sensing on the performance of secondary voice users is also taken into account. The QoS parameters of secondary voice users, which were obtained analytically, are also detailed. The analytical results were verified through the simulation, and will provide helpful insight in supporting voice services in a CRN.     

A Basic Dynamic Routing Problem and Diffusion

Diffusion theory has sometimes been successful in providing excellent approximate solutions to difficult queueing problems. Here we explore whether such methods can be used to analyze a basic dynamic routing strategy associated with a single idealized node in a data network. We analyze a dynamic routing policy where messages, or packets, that arrive at a certain node are routed to leave the node on the link having the shorter queue. In the model, message or packet arrivals are Poisson and the service time is exponentially distributed. We explore a heavy traffic diffusion method and we also discuss the limitations of an ad hoc approach to applying diffusion. For a node withKoutgoing queues we find, under the assumption of heavy traffic, the optimum dynamic strategy, in the sense of minimizing the average delay. When this optimum dynamic strategy is compared to a static strategy where the outgoing traffic is split among theKqueues, we find that the average delay for the dynamic system is better by a factor ofK.     

A performance model of deflection routing in multibuffer networkswith nonuniform traffic

Deflection routing can be used in networks whose stations have the same number of input and output links. Fixed length packets arrive synchronously on the station's input links at the beginning of time slots, and each packet is routed via the output link that offers the shortest path to its destination. Since the number of packet buffers at each output link is finite, the simultaneous contention of two packets for the last buffer of a common output link must be resolved by “deflecting” one of the packets to another output link. Thus, the deflection of a packet could result in the packet following a route that is not a shortest path. The potentially unbounded number of routes that a given packet can take makes analyzing the performance of such networks difficult. In particular, there are no analytical models that can analyze multibuffer deflection-routing networks with nonuniform traffic. Using independence assumptions, the authors develop a performance model of deflection routing that allows to estimate accurately and efficiently the mean transport time and throughput in a network that has any given two-connected topology, multiple buffers at each output port, and an arbitrary traffic matrix     

Delay constraint multipath routing for wireless multimedia ad hoc networks

According to the disadvantages of real time and continuity for multimedia services in ad hoc networks, a delay constraint multipath routing protocol for wireless multimedia ad hoc networks, which can satisfy quality of service (QoS) requirement (QoS multipath optimized link state routing [MOLSR]), is proposed. The protocol firstly detects and analyzes the link delay among the nodes and collects the delay information as the routing metric by HELLO message and topology control message. Then, through using the improved multipath Dijkstra algorithm for path selection, the protocol can gain the minimum delay path from the source node to the other nodes. Finally, when the route is launched, several node‐disjoint or link‐disjoint multipaths will be built through the route computation. The simulation and test results show that QoS‐MOLSR is suitable for large and dense networks with heavy traffic. It can improve the real time and reliability for multimedia transmission in wireless multimedia ad hoc networks. The average end‐to‐end delay of QoS‐MOLSR is four times less than the optimized link state routing. Copyright © 2014 John Wiley & Sons, Ltd.     

Joint admission/congestion control for wireless CDMA systemssupporting integrated services

A new data traffic control scheme is developed for maintaining the packet error rate (PER) of real-time voice traffic while allowing nonreal-time data traffic to utilize the residual channel capacity of the multi-access link in an integrated service wireless CDMA network. Due to the delay constraint of the voice service, voice users transmit their packets without incurring further delay once they are admitted to the system according to the admission control policy. Data traffic, however, is regulated at both the call level (i.e., admission control) and at the burst level (i.e., congestion control). The admission control rejects the data calls that will otherwise experience unduly long delay, whereas the congestion control ensures the PER of voice traffic being lower than a specified quality of service (QoS) requirement (e.g., 10 ^-2). System performance such as voice PER, voice-blocking probability, data throughput, delay, and blocking probability is evaluated by a Markovian model. Numerical results for a system with a Rician fading channel and DPSK modulation are presented to show the interplay between admission and congestion control, as well as how one can engineer the control parameters. The tradeoff of using multiple CDMA codes to reduce the transmission time of data messages is also investigated     

An extended least-hop distributed routing algorithm

A routing strategy called NELHNET has been developed for networks with multiprecedence traffic and operating under dynamic traffic and topological conditions. An adaptive distributed algorithm that uses least-hop and least-hop-plus-1 routes in a table of routing vectors, as opposed to the usual table of routing scalars, is described. Current delays are passed backward and forward with the packets to allow development of expected delays to each node via all acceptable routes. The route then selected is the acceptable route with the least expected delay. For speedier recovery, a node returning to service receives the current network status from an adjoining node as soon as the link connecting them is operational. The resultant algorithms show far greater than the marginal improvements originally expected over Arpanet simulations. NELHENET strategies also permit the network to function stably under more heavily loaded conditions than do the Arpanet strategies     

An integrated CDMA code assignment model for voice and data traffic in wireless communications

Wireless personal communication requires a provision of integrated services of multimedia traffic, such as voice and data, over the radio link. The multiple access protocols of code-division multiple-access (CDMA) techniques have been widely investigated in the recent literature. This paper presents an innovative multiple access protocol for CDMA-based wireless communication systems by fully utilizing the characteristics of voice and data traffic. In other words, a voice terminal can reserve a spreading code to transmit packets in multiple talk spurts, while a data terminal can transmit packets by either using the unassigned codes or borrowing the codes from the voice terminals during their silent periods. We build mathematical models for voice and data subsystems, respectively. Two performance parameters, the average dropping probability for voice packets and the average transmission delay for data packets, are derived based on the equilibrium point analysis. The effects of the two performance parameters on the system performance are discussed by varying the code reservation intervals of the voice terminals.     

Media Synchronization and QoS Packet Scheduling Algorithms for Wireless Systems

Wireless multimedia synchronization is concerned with distributed multimedia packets such as video, audio, text and graphics being played-out onto the mobile clients via a base station (BS) that services the mobile client with the multimedia packets. Our focus is on improving the Quality of Service (QoS) of the mobile client's on-time-arrival of distributed multimedia packets through network multimedia synchronization. We describe a media synchronization scheme for wireless networks, and we investigate the multimedia packet scheduling algorithms at the base station to accomplish our goal. In this paper, we extend the media synchronization algorithm by investigating four packet scheduling algorithms: First-In-First-Out (FIFO), Highest-Priority-First (PQ), Weighted Fair-Queuing (WFQ) and Round-Robin (RR). We analyze the effect of the four packet scheduling algorithms in terms of multimedia packet delivery time and the delay between concurrent multimedia data streams. We show that the play-out of multimedia units on the mobile clients by the base station plays an important role in enhancing the mobile client's quality of service in terms of intra-stream synchronization and inter-stream synchronization. Our results show that the Round-Robin (RR) packet scheduling algorithm is, by far, the best of the four packet scheduling algorithms in terms of mobile client buffer usage. We analyze the four packet scheduling algorithms and make a correlation between play-out of multimedia packets, by the base station, onto the mobile clients and wireless network multimedia synchronization. We clarify the meaning of buffer usage, buffer overflow, buffer underflow, message complexity and multimedia packet delay in terms of synchronization between distributed multimedia servers, base stations and mobile clients.     

Multipath Cognitive Routing in Integrated Multicast Satellite-HAP System Structures

A new multilayered inter satellite-high altitude platform(HAP)system routing algorithm is proposed,which is mainly based on multipath routing to ensure the network reliability.The proposed multipath routing scheme principally relies on splitting the traffic between different paths to make the best utilization of multiple routes.Linear programming is the main method used for multipath selection.The major constraints to the quality of service(QoS)(delay and link utilization)have been taken into account to meet the criteria of the advanced multimedia applications.Due to the effect of link utilization,the system encounters traffic flow oscillation between paths over time,which affects the system performance.Hence,to fix this issue,we propose a cognitive routing algorithm which reacts to the long-term changes of the traffic loads rather than short-term ones.The performance of the proposed routing techniques has been evaluated using appropriate simulation models and implemented by Matlab.     

Road Aware Geographical Routing Protocol Coupled with Distance,Direction and Traffic Density Metrics for Urban Vehicular Ad Hoc Networks

The new information and communication technologies have changed the trend of communication in all fields. The transportation sector is one of the emerging field, where vehicles are communicating with each other or with infrastructure for different safety and comfort applications in the network. Vehicular ad hoc networks is one of the emerging multi-hop communication type of intelligent transportation field to deal with high mobility and dynamic vehicular traffic to deliver data packets in the network. The high mobility and dynamic topologies make the communication links unreliable and leads to frequent disconnectivity, delay and packet dropping issues in the network. To address these issues, we proposed a road aware geographical routing protocol for urban vehicular ad hoc networks. The proposed routing protocol uses distance, direction and traffic density routing metrics to forward the data towards the destination. The simulation results explore the better performance of proposed protocol in terms of data delivery, network delay and compared it with existing geographical routing protocols.

     

Iterative approach to optimizing convergence routing priorities

This paper shows how to optimize the routing decisions in a nondeterministic routing algorithm called convergence routing in which routes may change depending on the traffic conditions. The routing algorithm guarantees a loss-free delivery of data packets from bursty sources, and a deterministic bound on the route length in arbitrary topology networks. The routing decisions are based on assigning routing priorities to the links such that a packet is forwarded to the highest priority link which is not blocked. Routing priorities are assigned using a local-greedy metric which minimizes the distance (number of hops) to the destination. This work shows that routing decisions using a local-greedy metric are not optimal, and the performance of the algorithm can be improved substantially by using new measures. Thus, various look-ahead metrics which take into account the potential gain on the other switching nodes toward the destination of a packet are suggested. The contributions of this work are: (1) a new analytical model to capture the behavior of a switching node; (2) an iterative optimization technique to set routing priorities according to various look-ahead measures; and (3) heuristics to ensure the stability of the routing priorities. The optimization objective is to maximize the throughput by minimizing the maximum total flow carried on a link in the network under static traffic model. The performance is studied computationally on various networks and traffic matrices. It is shown that up to a 50% performance increase can be obtained by optimizing the routing priorities     

Routing Optimization in IP Networks Utilizing Additive and Concave Link Metrics

Routing optimization provides network operators with a powerful method for traffic engineering. Its general objective is to distribute traffic flows evenly across available network resources in order to avoid network congestion and quality of service degradation. In this paper we consider routing optimization based on conventional routing protocols where packets are forwarded hop-by-hop in a destination-based manner. Unlike other work in this area, we consider routing protocols, which are able to take into account concave routing metrics in addition to additive ones. The concave link metric introduces an additional degree of freedom for routing optimization, thus, increasing its optimization potential. We present and evaluate a mixed-integer programming model, which works on these metrics. This model unifies the optimization for single-metric and dual-metric routing concepts and also includes the consideration of multipath routing. Furthermore, we propose a heuristic algorithm usable for larger network instances. Numerical results indicate that employment of both the dual-metric concept and multipath routing can achieve considerably better utilization results than default-configured single-metric routing. A significant finding is that metric-based routing optimization with two link metrics often comes close to the results obtainable by optimization of arbitrarily configurable routing.     

Improved multicast routing in MANETs using link stability and route stability

The growing popularity of multimedia applications and services needs to support several quality of service metrics such as high throughput, low energy, and jitter, which is a challenging task in mobile ad hoc networks. Because of limited bandwidth, energy constraints, dynamic topology, transmission errors, and fluctuating link stability, the links between adjacent nodes are often not reliable and may break because of node mobility. Link breakage initiates the process of rerouting either at the sender node (the node at which the link breaks) or at the source node. In either case, it leads to packet loss, delivery delays, and increased control overheads. Hence, to attain a minimum quality of service, routing protocols must address the dynamic network topology. Uncertain and varying movement of nodes necessitates stability of the links between such nodes. The objective of this paper is to propose 2 protocols, the first based on link stability and the other based on route stability. Link stability identifies a stable link from the available links to the next hop and determines a stable end‐to‐end route. The probability of successful transmission of periodic packets is used as a link stability metric to assess the stable path. Acknowledgment ‐free packets are used to check connectivity in the network. Increased probability of successful transmission implies that the selected link is sustained for longer duration and can deliver packets more reliably or, as a consequence, results in a stable link to deliver a better data rate. With a stable link, there is a reduced possibility of retransmissions, reduced end‐to‐end delay, reduced control overheads, and enhanced data delivery ratio. Selection of the most stable route for data transmission improves the performance. Experimental results from simulations performed on EXata/Cyber v2.0 simulator reveal that our proposed protocols are an improvement over the existing protocols in terms of packet delivery ratio, average end‐to‐end delay, and average route lifetime, even without route optimization with the minor increase in control packets. A case study of the application of proposed protocols is also presented.