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Being increasingly dense and mature in layout, Internet has become an infrastructure for a state. The traditional single server can no longer bear the network traffic in rapid growth. For example, once the booking service for concert tickets is made available to the public, we are always prompted when clicking into the booking system that the server is busy. And when the system is finally restored normal, the tickets have been sold out. The load issue puzzling the ticket system also occurs in the transportation section every time when a long holiday is coming. Even though the system is updated on a yearly basis, net- work congestion still happens frequently. Therefore, there is still great space for the development of the ways and mechanisms of load balance in traditional multi-servers. Evaluation of the mech- anism and efficacy of load balance through P4Switch is proposed in this paper in the hope to achieve more efficient server load balance.

A following vehicle can significantly reduce fatigue in people who want a hands-free walk. The vehicle follows a person and carries heavy things for him/her so that the person can walk in comfort without the burden of carrying the heavy things. Robot Operating System, ROS, is a software framework which is widely adopted for the development of automated guided vehicles, AGVs. Following vehicles and AGVs share many similarities. Both of them can move goods and autonomously plan their paths without manual intervention. Thus, this paper proposes a following vehicle prototype utilizing ROS. In a ROS robot, we integrate a program to measure the power present in received radio signals from a radio transmitter attached to the person it follows. The measurements indicate the approximate distance between the following vehicle and its user. The vehicle, then, moves toward the user based on the estimated distance. The experiment results prove that the proposed prototype can correctly follow the movement of its user in a linear direction with an obvious but acceptable delay.

With the continuous development of network technologies, a growing number of devices are connected to the Internet, and the reliability, availability and data transmission performance of the network have become increasingly important. Software-Defined Network (SDN) breaks the closed system of the traditional network that is formed by the tight coupling among network hardware, operating system and applications, enabling the network to be elastically adjusted based on different environments and needs. In this study, a mechanism for improving the reliability, availability and transmission performance of network was built in the SDN environment. In the proposed approach, the stochastic mechanism was applied to distribute traffic and the OpenFlow Group Tables were used to enable the automatic switch to the standby Action Bucket for uninterrupted service in case of any fault of the link so as to improve the fault tolerance and reliability of network. Experiments demonstrated that the proposed mechanism can transmit data faster and achieve more reliable network services.

This study investigated programmed control of transmitting video and audio streaming data with P4 switches in software defined networks. When a packet is transmitted to a P4 switch, the switch checks the match actions and invokes a random function to determine which output port the packet should be sent to. Thus, the switches in the network can route the streaming data via different path according to the random function so as to provide multipath transmission. Simulation results showed that the proposed multipath transmission mechanism can improve transmission efficiency, reliability, and network throughput.

Software Defined Networking is a new networking paradigm that has emerged recently as a promising solution for tackling the inflexibility of the classical IP networks. The centralized approach of SDN yields a broad area for intelligence to optimise the network at various levels. Fault tolerance is considered one of the most current research challenges that facing the SDN, hence, in this paper we introduce a new method that computes an alternative paths re-actively for centrally controlled networks like SDN. The proposed method aims to reduce the update operation cost that the SDN network controller would spend in order to recover from a single link failure. Through utilising the principle of community detection, we define a new network model for the sake of improving the network's fault tolerance capability. An experimental study is reported showing the performance of the proposed method. Based on the results, some further directions are suggested in the context of machine learning towards achieving further advances in this research area.

Nowadays, the demand for data transmission is ever increasing with the increase of network applications. Therefore, how to enhance the performance of data transmission is an issue worthy of in-depth discussion. The throughput of single-path transmission schemes might be throttled by a congested link of its path. In contrast, multipath transmission schemes can utilize the bandwidth resource of several disjoint paths to increase the throughput. In this paper, we proposed a multipath transmission scheme to equally balance the traffic of a flow over the disjoint paths designated by the Software Defined Network (SDN) controller, which is able to learn the topology of the network and discover multiple transmission paths. Simulations were conducted with Mininet to evaluate our proposed transmission scheme. In comparison with the traditional single-path transmission scheme, which always adopts the shortest path, experimental results show that our proposed scheme achieves a significant improvement on performance in terms of transmission throughput.

This paper develops a hierarchical packet pre-dropping (HPPD) approach that exploits different methods to process I (intra-coded)/P (predictive-coded) /B (bidirectionally predictive-coded) video frame packets to improve the video transmission quality over IEEE 802.11e Wireless Local Area Networks (WLANs). HPPD refers to the dropping of arriving packets if the preceding video frame packets in the same Group of Pictures (GOP) have been dropped. The simulation results demonstrate that, in terms of PSNR, the proposed scheme outperforms exiting schemes such as EDCA (Enhanced Distributed Channel Access) and adaptive mapping and novel mapping mechanism, suggesting that the resource of access channel in WLANs is not wasted in the transmission of useless video data.

A stream-based mapping mechanism (SBM) is proposed for improving the quality of MPEG-4 video delivery. The proposed design keeps two factors in our previous works: significance of video frames and network traffic loads. In addition, SBM considers the action taken on the packet of the same video frame transmitted previously to improve the quality of video transmission over IEEE 802.11e networks. The proposed scheme is evaluated in networks with different traffic loads. The simulation results show that, in terms of frame loss ratio and PSNR (Peak to Signal Noise Ratio), the proposed scheme outperforms exiting schemes such as EDCA (Enhanced Distributed Channel Access), static mapping and adaptive mapping.

Unequal error protection (UEP) on streaming media broadcasting is a widely used technique over wireless environment. Forward error protection (FEC) with UEP is a prevailing scheme to improve the quality of video streaming over lossy channels. Frame-level FEC have been proposed for video streaming due to the priorities of video frames within transmission constraint on Bernoulli channels. However, in current Internet architecture, various communication and storage systems are prone to corrupt as burst of noises occurs. The corrupted data pieces may exceed the error correction capacity of FEC, and therefore degrade the efficacy of FEC. To address this problem, we proposed a model to evaluate the perceived quality of H.264/AVC video streaming over burst channels. The experiment results indicate that the proposed scheme is able to evaluate the video streaming quality in perspective of decodable frame rate (DFR) with lower complexity than alternative methods.

Robotic devices are increasingly a research focus for liberation of humans from time-consuming traditional tasks, but the cost and complexity of such devices remain a daunting issue. This paper proposes a simple and cost-effective demonstration prototype system composed of a wireless module, sensors and a mobile robotic cart which prowls the target environment, for example a home or professional healthcare room for the elderly. Using low cost off-the-shelf components, the presented system is configured as a fire detector/alarm/extinguisher, but is adaptable to many other configurations. It demonstrates that, with adequate planning and forethought, sophisticated robotic performance at low material and developmental cost can be obtained for common household, healthcare and industrial applications.

Video streaming over wireless networks is a challenging task due to high packet loss rate, low available bandwidth, and burst packet losses problems. Accordingly, this paper extends our previous works to propose Forward Error Correction with Interleaving mechanism combining Cognitive Technology (FECI-CT) for video streaming over wireless networks. FECI-CT uses an FEC scheme which recovers the lost packets to overcome the high packet loss rate and calculates the minimum error recovery overhead which reduces the bandwidth consumption in transmitting FEC redundancy. The interleaving scheme transforms burst packet losses into uniform packet losses to mitigate the burst packet losses problem. The Cognitive Technology (CT) dynamically seeks an unused channel for transmission in order to increase the available wireless bandwidth. The simulation results show that the FECI-CT mechanism can obtain satisfactory video quality for video streaming over wireless networks.

The current IEEE 802.11 Distributed Coordination Function (DCF) that provides equal medium access probability to all transmitters suffers from uplink and downlink imbalanced link problem when operating in the infrastructure mode. Therefore, this paper proposes an Enhanced High-performance Distributed Coordination Function (EHDCF) to mitigate this imbalanced link problem. In EHDCF, the current transmitting station selects the access point (AP) with higher probability and the wireless stations with lower probability as the next transmitter to improve the downlink throughput. Simulation results show that EHDCF not only provides fair bandwidth sharing for uplink and downlink traffic but also yields a significant improvement in the network throughput.

In this study, we propose an efficient localization scheme in wireless sensor networks. The proposed scheme utilizes mobile anchors and is based on ring overlapping. In a wireless sensor network, there are some nodes knowing their locations, which are called the reference nodes, and the other nodes without the knowledge of their locations are called the blind nodes. Based on the relative RSSI (Received Signal Strength Indicator) values, we utilize mobile beacons to find out the rings that are centered with a certain reference node and contain the blind node. These rings are called B-Rings. Since the mobile anchor and the reference node know their own locations, the B-rings can be precisely derived. By utilizing the mobile beacons, the widths of the B-Rings can be further minimized; and then by overlapping them, the location of the blind nodes can be efficiently estimated. Besides, most existing location schemes that utilize mobile anchors let the mobile anchor move randomly. In contrast, we propose a regular and simple movement scheme for the mobile anchor. Thus, the mobile anchor consumes less energy than other schemes in which the mobile anchor move randomly. Simulations of the proposed scheme show that the proposed localization mechanism has better accuracy than other existing related approaches

Vehicular wireless networks have emerged as a promising wireless communication technology in recent years. Vehicular wireless networks collect information on real-time traffic, such as the road situation, the location of a car accident, etc., which must be delivered to its destination before a set deadline in order to be useful for an Intelligent Transportation System (ITS). Target applications are called safety-critical information. However, vehicular wireless communication does not incorporate reliable transmission of safety-critical information which will experience wireless errors due to collisions or the noisy environment caused by the high speed of moving vehicles. If the driver of a vehicle does not receive the information in time, this may cause a critical condition. This paper proposes a Reliable Transmission Mechanism (RTM) to solve this problem. The RTM relies on extra parity bytes adaptively added to a byte-level forward error correction mechanism to enhance the transmission quality of safety-critical information. This paper simulates the RTM and compares it with conventional vehicular wireless networks in terms of effective packet loss rate, retransmission count and average end-to-end delay. The simulation results show that our proposed RTM protects the information effectively.

In recent years, the issue of Vehicular Ad-Hoc Network (VANET) has received great attention, and more and more VANET-related researches have been brought up. Generally speaking, the biggest difference between VANET and traditional Ad-Hoc is the velocity of carriers because in VANET, the velocity of vehicles the carriers is much higher than that of the carriers in traditional Ad-Hoc. Furthermore, it would be a great challenge to forward data efficiently in VANETs. Thus, many researches proposed have focused on the development of Routing Protocols. The main purpose of this paper is to investigate Geographic Position-Based Routing Protocol, which needs Global Position System (GPS) to obtain the positions of vehicles and to enhance the decision-making of data delivery. Therefore, only partial topology information is needed during the decision-making of data delivery for the routing protocol to have better adaptability in high-speed network topology. This paper aims to improve Greedy Perimeter Stateless Routing (GPSR) Protocol and implement our proposed scheme into a complex scenario like urban area. For this reason, with the concept of vector, greedy forwarding mode of GPSR not only concerns the distance to enhance the accuracy of routing data while choosing the next hop, but also, in terms of the intersections in urban area, adds a predictive mode to predict the motions of vehicles at the intersections to improve the efficiency of routing protocol.

To prevent decoding failure of MPEG-based applications caused by error propagation, the present study modifies multi-streaming mechanism of SCTP protocol and proposes a transmission control approach which can adjust transmission policy dynamically according to the quality of service requirement of individual frame. The simulation results confirmed that the proposed scheme can improve the performance of MPEG-based video stream by modified SCTP multi-streaming mechanism.

Based on the standardized IEEE 802.11 Distributed Coordination Function (DCF) protocol, this paper proposes a new backoff algorithm, called Smart Exponential-Threshold-Linear (SETL) Backoff Algorithm to enhance the system performance of contention-based wireless networks. As we know, the smaller contention window (CW) will increase the collision probability, but the larger CW will delay the transmission. Hence, in the SETL scheme, a threshold is set to determine the network load. When the CW is smaller than the threshold, a light network load, the CW size is self-adjusted exponentially. Conversely, if the CW is larger than the threshold, a heavy network load, the CW size is tuned linearly. In addition, the SETL takes a more conservative measure by decrease the CW after “S” times consecutive successful transmission to reduce the collision probability, especially when the competing station is large. By simulation, the numerical results show that the SETL provides a better system throughput and collision rate in both light and heavy network load than the related backoff algorithm schemes, including binary exponential backoff (BEB), exponential increase exponential decrease (EIED) and linear increase linear decrease (LILD). The SETL is very easy to implement, as it dose not require any changes in DCF procedures. Every station will self-adjust CW well with high performance and low collision rate.

In recent years, voice over Internet Protocol (VoIP) has gained much attention in wireless local area networks (WLANs). Although IEEE 802.11e has been standardized to provide voice traffic with quality of service (QoS) support, the voice quality will still be severely degraded with a large amount of best-effort traffic generated by distributed coordination function (DCF) or enhanced distributed channel access (EDCA) stations. In this paper, we propose a novel ACK mechanism called delay-based ACK (DB-ACK) scheme to protect voice traffic from legacy DCF users and AC_BE traffic by using adaptive DIFS and AIFS values at DCF and EDCA stations respectively in an 802.11e infrastructure network. We evaluate the performance of the DB-ACK scheme through extensive simulations. Simulation results show that the proposed scheme can well support the stringent QoS requirements for voice traffic while maintaining acceptable throughput for best-effort traffic.

This paper addresses the QoS support issue of IEEE 802.11e in multirate networks. Prior researches focused on the "performance anomaly" problem without QoS consideration. We show the QoS support is insufficient in the current 802.11e standard and related methods make it worse by using NS-2 simulation tests. We propose a hybrid method that maintains the quality of highest priority traffic and enhance the system performance concurrently. We also compare its performance with other proposed methods. Our results demonstrate that our proposed hybrid method can efficiently support the time constraints of voice flows in various network conditions. Moreover, the system throughput is improved by the hybrid method when compared to the IEEE 802.11e standard.

Fading is one of the common characters of a wireless channel. Many rate adaptation mechanisms were proposed to enhance the IEEE 802.11 systems, but these may cause multiple rates in a single network. However, IEEE 802.11 DCF faces the performance anomaly in multi-rate network, that is, when the stations with different rates collide, the low-rate station will greatly degrade the high-rate station. In this paper, we propose a collision avoidance multi-rate MAC (CAMR-MAC) protocol, which is different from the current solutions for performance anomaly. Current performance anomaly solutions are to assign different priorities to stations with different rates based on the IEEE 802.11 DCF protocol, which is still a fully contention based protocol. The simulation results show that CAMR-MAC not only improves the network throughput but also effectively guarantees the temporal fairness of all stations.

Online monitoring is an emerging field comprising the enhancement of rural development in the areas of education through improved information exchange, communication and learning processes, based on the use of Internet and other digital technologies by actors locally, regionally and worldwide. Today online monitoring over IP network rely on an IP camera and network infrastructure. IP camera comes with a high resolution CCD sensor that provides a sharp and clear image with life-like color representation but the IP network reliability and bandwidth are crucial for the development of online monitoring system as well as an efficient use of the available resources in rural areas. The paper suggests the way of managing the bandwidth by reducing or eliminating unidentified network traffic of an organization to ensure networkpsilas uptime, reliability, and speed for the efficient use of the network.

Video streaming over wireless networks have many challenges due to the high error rate and burst packet error characteristic. Forward error correction (FEC) is a method commonly used to handle losses in real-time communication. Conventional FEC mechanisms provide redundancy by an averaged packet loss rate and performance decreases by burst packet losses. However, the average packet loss rate cannot give any indication of burst packet loss. Hence, the conventional FEC mechanisms cannot recover original source data over wireless networks. In this paper, we propose a burst-aware adaptive FEC (BAFEC) control mechanism to overcome burst packet losses. We will therefore be able to take account of average burst packet loss length. The sender can rely on this information in order to adjust the FEC redundancy. The experimental result shows that compared to the conventional FEC mechanisms, our proposed mechanism achieved better recovery performance in terms of packet loss rate and PSNR.

Nowadays, multimedia applications are expected to be an important traffic source in wireless networks. Such sources usually have stringent delay requirements and in many cases they should be delivered exactly as they were generated. In this paper, we propose a novel pulling based uplink packet scheduling policy for rtPS/ertPS (real-time polling service/extended real-time polling service) traffic to support real-time applications, with strict delay requirements, such as VBR (variable bit rate) traffic in IEEE 802.16e BWA (broadband wireless access) systems. The proposed transmit-permission policy can derive sufficient conditions such that all the rtPS/ertPS sources satisfy their time constraints to provide deterministic QoS (quality of service) guarantees. In addition to theoretical analysis, simulations are conducted to evaluate the performance of the proposed scheme. As it turns out, our design provides a good quality performance in the IEEE 802.16e BWA systems.

Use of a variety of multimedia applications in Internet usage has grown up rapidly in recent years. This has stimulated the study of networking mechanisms for providing QoS guarantees. Simulators play an important role when researchers want to validate their proposed networking mechanism. In order to pursue more realistic simulations, some researchers devoted themselves to the improvement of simulators resulting in some simulation frameworks based on NS-2 and other programs. In this paper, to achieve more realistic simulations, we combined MyEvalvid_RTP with NS-2 to develop a new simulation tool-set.

Quality of service (QoS) support in multimedia applications is a crucial requirement. Many researchers focus on the performance enhancement of video transmission and they use simulation tools to evaluate their proposed mechanisms. Most simulation tools only use network-level performance metrics, such as throughput. However in some cases, network-level performance metrics can not evaluate the video or audio delivered quality well. To address this issue, a series of simulation tool-sets are developed. However, current simulation tool-sets can only evaluate the video delivered quality; they cannot evaluate the audio delivered quality. Therefore, to achieve more realistic simulations, we combined Evalvid with NS-2 VLC and RTPtools to obtain a new tool-set called as MyEvalvid_RTP. By using this new simulation tool-set, researchers can evaluate not only the video delivered quality but also the audio delivered quality.

In this paper we present a new framework for quality H.264 video delivery from wired to wireless networks. The proposed solution is an extension of ETBMS in conjunction with IEEE 802.11e wireless protocol in EDCF mode. The analysis of network delay and video quality shows that the proposed solution which redefines the DSCP fields in DiffServ network, uses AF mechanism with IEEE 802.11e protocol for H.264 video traffic, yields significant results. Thus a video data can be provided with better Quality of Service (QoS) in a heterogeneous network.

The QoS support for multimedia on wireless network is one of the most important researches in recent years. IEEE wireless 802.11e is the most hopeful among these methods. However, the performance metrics used in 802.11e experiments are almost network-level measures. They represent the network state and not the perceptual quality of the end user. Besides, the wireless network is error prone and errors are unavoidable. In this paper, we evaluate the effect of random error model and Gillbert-Elliot (GE) error model on the quality of multimedia applications.

In this paper, we propose a novel realistic simulation tool-set known as MyEvalvid-NT for evaluating video delivery over simulated networks. This tool-set integrates Evalvid and NS2. Though PSNR is most often used to measure the perceived quality by an end user, it is time-consuming to calculate PSNR when evaluating a long video sequence. To avoid overhead, in this paper, we used decodable frame rate (Q) as our performance metric to evaluate the received video quality. We use a video transmission over erroneous wireless network to demonstrate the simulation using this tool-set. We also showed that Q can measure perceived video quality adequately as PSNR. With this tool-set, researchers can easily analyze their designed mechanisms, such as network protocols or QoS control schemes in a realistic simulation environment. In brief, researchers who utilize our proposed QoS assessment framework will be benefited in verifying their designs regarding video transmission over simulated network.

The purpose of this paper is to study the packet loss effect on MPEG video transmission quality in wireless networks. First, we consider the distribution of packet losses in wireless network, including distributed and burst packet losses. Besides, we also discuss the additional packet drops by the play-out buffer at the received end, and the effect of the transmission packet size on the video delivered quality. From the results, we find that the effect of burst packet losses on the video delivered quality is less than distributed packet losses in the same packet loss rate. Moreover, the smaller size of the play-out buffer leads to more packet drops and worse video quality. Finally, if there is no video recovery for video transmission, the video delivered quality of the larger packet size will be better than smaller packet size.

The need for QoS for multimedia traffic arouse from the fact that the current network is a best effort network where all packets are treated equally irrespective of their importance. Moreover when a video packet changes its mode of transport from wired to wireless there arises a need to identify these packets and provide them with prioritized service to achieve end-to-end QoS. In this paper we used a DiffServ network model for the wired network where the INGRESS executed enhanced token bucket three marker system (ETBMS). The framework also describes a video differentiator present in the access point which identifies the video packets and maps it to corresponding IEEE 802.11e prioritized video service queue. We demonstrated via simulation the proposed concept and examined the QoS parameters like end-to-end delay and video quality. The results illustrate that the proposed scheme exhibits better video delivery and QoS in heterogeneous networks.

This paper proposes an enhanced adaptive FEC (EAFEC) algorithm implemented in the access point (AP) to improve video delivery over wireless networks. Unlike traditional static FEC mechanisms, which add redundant data to transmission data in a fixed number, the number of redundant FEC data for the EAFEC mechanism is determined by the AP, which is based on both network traffic load and wireless channel state. The algorithm is based on two factors. One of them is the queue length in the access point, indicating network traffic load; the second factor is packet retransmission times, indicating wireless channel state. EAFEC usage would tune FEC data number generated in such a way that it would not unnecessarily add more traffic to the network in already congested scenario. Our work shows that the EAFEC algorithm improves system performance by dynamically tuning FEC strength to the current loss in the wireless channel. We use NS-2 simulation experiments to prove our algorithm

Multimedia wireless network QoS support has become one of the most important researches in recent years. IEEE wireless 802.11e is the most hopeful among these methods. However, 802.11e experiment performance metrics are essentially network-level measures. They represent the network state, not the perceptual quality of the end user. This paper presents an application-level study on the wireless 802.11 DCF and 802.11e EDCA scheme using an NS-2 simulator.

The paper presents an implementation of QoS library redirection (QLR) for DiffServ in Microsoft Windows systems. In the QLR method, the legacy application obtains packet pre-marking ability without any modification of source codes. We have upgraded the FTP and vic (video conferencing) applications by using QLR in our Microsoft Windows testing platform. Our experimental results show that this approach can use really effectively and provide high performance of the QLR in Microsoft Windows systems.