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Data center networks (DCNs) are the backbone of many cloud and Internet services. They are vulnerable to link failures, that occur on a daily basis, with a high frequency. Service disruption due to link failure may incur financial losses, compliance breaches and reputation damage. Performance metrics such as packet loss and routing flaps are negatively affected by these failure events. We propose a new Bayesian learning approach towards adaptive path allocation that aims to improve DCN performance by reducing both packet loss and routing flaps ratios. The proposed approach incorporates historical information about link failure and usage probabilities into its allocation procedure, and updates this information on-the-fly during DCN operational time. We evaluate the proposed framework using an experimental platform built with the POX controller and the Mininet emulator. Compared with a benchmark shortest path algorithm, the results show that the proposed methods perform better in terms of reducing the packet loss and routing flaps.

An advanced metering infrastructure (AMI) consists of a meter data management system, a concentrator, and a meter. Each component can communicate with all other components, through either wired or wireless connections. In an AMI local area networks, various protocols, such as Zigbee, WiFi, and power-line communication, have the potential to be used in AMI communication. Of these protocols, Zigbee offers self-constructing networks, low costs, and superior safety levels. Zigbee is adopted as the LAN communication protocol because of these advantages in this study. In a LAN, a concentrator may encounter many emergent situations, any of which might cause the concentrator to lose its network connection function. Consequently, additional time is required for network construction, which tends to reduce the reliability of smart grids with LANs of this type. To solve these issues, this paper proposes the fast join process (FJP) and enhanced FJP to shorten the time of Zigbee network construction. The strategy employed in this paper is to let every meter join a specified concentrator. The proposed FJP does not scan unnecessary channels. By simplifying association message exchange, we further propose a new method called the enhanced FJP. By using these methods, not only can message flow be managed easily, but the time of network construction can also be improved.

In recent years, the emerging paradigm of software-defined networking has become a hot and thriving topic in both the industrial and academic sectors. Software-defined networking offers numerous benefits against legacy networking systems by simplifying the process of network management through reducing the cost of network configurations. Currently, data plane fault management is limited to two mechanisms: proactive and reactive. These fault management and recovery techniques are activated only after a failure occurrence and hence packet loss is highly likely to occur. This is due to convergence time where new network paths will need to be allocated in order to forward the affected traffic rather than drop it. Such convergence leads to temporary service disruption and unavailability. Practically, not only the speed of recovery mechanisms affects the convergence, but also the delay caused by the process of failure detection. In this paper, we define a new approach for data plane fault management in software-defined networks where the goal is to eliminate the convergence process altogether rather than accelerate the failure detection and recovery. We propose a new framework, called Smart Routing, which allows the network controller to receive forewarning signs on failures and hence avoid risky paths before the failure incidents occur. The proposed approach aims to decrease service disruption, which in turn increases network service availability. We validate our framework through a set of experiments that demonstrate how the underlying model runs and its impact on improving service availability. We take as example of the applicability of the new framework three types of topologies covering real and simulated networks.

Recently, increasing attention has been paid to the study of applying software defined networks to improve video transmission. Compared with the existing studies which focus on a wired network, this study proposed a multi-path transmission mechanism for improving the performance of transmitting videos from a wired network to a wireless one in a software defined network environment. In the proposed mechanism, a mobile host could apply network bonding to integrate several physical or virtual wireless network modules, each of which was connected to a different access point, into a combined device. Thus, multiple wireless links of the mobile host could be established and hence multiple transmission paths from the video source in the wired network to the mobile host were derived. In addition, different amount of video packets were allocated onto different transmission paths so as to maximize the performance of the multi-path transmission. According to the results of our experiments, the proposed mechanism can enhance the performance of transmission in terms of Peak Signal to Noise Ratio (PSNR).

The increasing complexity of modern day networked applications and the massive demand on the Internet resources has reignited interest and concern in the underlying networking infrastructures and their ability to cope with such complexity and adapt to the demands of the business applications, particularly where such applications require a high degree of robustness and reliability. As a result, software-defined networking has emerged as a promising approach to the definition of network architectures that could carry a high degree of adaptability and robustness reminiscent of the future Internet. Fault tolerance and network updates are considered two of the current research challenges that hamper the growth of software-defined networking in this area. Therefore, this paper represents a step toward tackling these two issues in the context of single link failures. Our main contribution lies in the definition of new algorithms that aim to enhance the problem of finding alternative paths in large-scale networks with minimal cost and time-to-update factors. The new solution aims at increasing the efficiency of flow operation reduction during link failures. We evaluate our framework and show how its implementation results in improved efficiency.

Packets that traverse more hops to the destination will get lower throughput and result in the inter-flow fairness issue.We propose a virtual queue management scheme that does not require the modification of any communication protocol.According to the number of flows, our scheme adjusts the parameters to achieve each flow's fair sharing of channel resource.Our proposed scheme can mitigate the inter-flow fairness problem and effectively improves the quality of video transmission. Due to intra-flow and inter-flow interference problems, the throughput performance decreases dramatically in a multi-hop wireless network. These two kinds of bandwidth unfair sharing problems could cause serious collisions and congestion, hence affecting the performance of multi-hop wireless networks. That is to say, data packets that need to traverse more hops to arrive at the destination will get lower throughput and result in the inter-flow fairness problem. Furthermore, the quality of video transmission is especially poor in traditional multi-hop wireless network environments. Therefore, in this paper, we propose a virtual queue management scheme that does not require the modification of any communication protocol. According to the number of flows, it adjusts the queue management scheme to achieve each flow's fair sharing of channel resource. It also improves the quality of video transmission. Through NS2 simulations, the results show that our proposed scheme can mitigate the inter-flow fairness problem and effectively improves the quality of video transmission. Display Omitted

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.

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.

In this study, we develop a hierarchical packet pre-dropping (HPPD) approach that exploits different methods to process intra-coded (I)/predictive-coded (P)/bidirectionally predictive-coded (B) video frame packets to improve the video transmission quality in heavily loaded IEEE 802.11e wireless local area networks (WLANs). It is not easy to transmit packets in WLANs with a heavy traffic load. Efficient transmission of video packets is important for improving the quality of video transmission. If the network becomes congested, important packets may be dropped. Further, video packets tend to be affected considerably more than data packets. Therefore, controlling the traffic of insignificant packets to avoid the transmission of useless data can improve the quality of video transmission. On the basis of this concept, in this paper, we propose a strategy that is different from the approaches proposed in earlier studies. First, identification of the dependence of video frames in a group of pictures (GOP) can prevent the transmission of useless data. Thus, the transmission channel resources are economized by avoiding the transmission of undecodable video frames. Second, in heavily loaded WLANs, it is not sufficient to utilize the queues to transmit important video frame packets. HPPD adopts the method of an aggressive drop of unimportant packets to retain the transmission channel resources for important video packets. Experiments have demonstrated that in heavily loaded WLANs, the Peak Signal to Noise Ratio (PSNR) for the case when the HPPD mechanism is adopted is better than that when enhanced distributed channel access (EDCA), adaptive mapping, or a novel mapping mechanism is used.

The nodes in wireless sensor networks (WSN) are typically resource constrained so that they can maintain only a few routes. More-capable nodes can insert extra routing information, i.e. source routing header (SRH), into packets to instruct the resource constrained ones to route the packet. A WSN with deeper depth requires longer SRH, thus leaving less space of a packet for the user data. We analyze the relationship between the length of the user data and the maximum depth of a WSN based on RPL and IEEE 802.15.4. The results can guide the application designers and the network administrators in selecting a suitable length of user data to guarantee that the data can be routed to each sensor nodes. Simulation results prove the correctness of our analysis.

IEEE 802.11e has been introduced to support quality of service (QoS) in WLANs by allocating different traffics to four access categories. Cross-layer design is a method to improve the transmission quality of multimedia over wireless networks. However, when the fragmented packets of a single frame are allocated to different AC queues, out-of-order delivery may occur. Then the receiving node requires more buffer space to absorb the out-of-order packets. Therefore, even with a significance classification mechanism, the quality of the video transmission may still be limited due to the occurrence of out-of-order delivery and the transmission of information that becomes useless to the receiver. Therefore, this work developed a novel mapping mechanism (NMM) to improving the quality of MPEG-4 video delivery over IEEE 802.11e networks. The NMM considers the action that was taken on packet of the video frame that had been previously transmitted to resolve the out-of-order delivery by allocating them to the same AC queue as the previously process if possible. Furthermore, the proposed mechanism NMM takes more aggressive scheme to protect more important frame packets from being dropped. In order to evaluate the quality of MPEG-4 video delivery over IEEE 802.11e networks by the proposed mechanism, this work adopted two evaluation metrics, which is called Peak to Signal Noise Ratio (PSNR) and the ratio of frames loss. The simulation results demonstrate that, in terms of both PSNR and the ratio of frames loss, the proposed scheme outperforms exiting schemes such as EDCA (Enhanced Distributed Channel Access) and adaptive mapping.

Wireless networks have a varying channel quality and a limited bandwidth. Therefore, ensuring a satisfactory video quality over wireless networks is extremely challenging. Most existing studies for enhancing the received video quality over wireless networks consider only single-hop transmission environments. Consequently, when applied to multi-hop wireless networks, such as wireless mesh networks, VANETs or wireless sensor networks, the video quality is typically rather poor. Accordingly, this study proposes a cross-layer mechanism for improving the video transmission quality in multi-hop wireless networks. The proposed scheme considers both the video frame type and the output interface queue length at the intermediate nodes while determining the drop probability during the packet forwarding process. Besides, in the event of transmission failures, the number of retransmission attempts is varied adaptively in accordance with the network conditions and the relative priority of the corresponding frame. Furthermore, the forwarding probability at each node is increased in accordance with the number of hops previously traversed by the packet. Thus, the proposed scheme improves video delivery quality by means of providing unequal protection for frames of different importance. The NS2 simulation results show that the average PSNR results obtained with the proposed scheme have significant improvements compared to that obtained with the traditional IEEE 802.11 mechanism, thereby confirming the effectiveness of the proposed scheme.

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.

Congestion control in Vehicular Ad Hoc Networks (VANETs) is challenging due to limited bandwidth, dynamic topology and lack of central coordination. Unlike previous works which focus on congestion control with a fixed channel load threshold that causes bandwidth wastage, this paper presents a cross-layer congestion control model which consists of two modules to alleviate congestion in the congestion detection center. In first module, the event-driven messages are prioritized when an imminent danger or abnormal situation is detected on the road. Then in second module, the channel load threshold is assigned dynamically based on beaconing load and transmit power properties which results a satisfaction level of maximum load beaconing. As a result, it alleviates congestion problems and improves bandwidth usage in VANETs. Experimental results on different data-sets including various vehicle densities, distances and road maps with and without obstacles (e.g. walls and buildings) show that the proposed method outperforms existing methods in terms of the average delivery ratio, message reception probability and average delay of time. Moreover, the results prove that the performance of all approaches degraded significantly in realistic scenarios (i.e. scenario with obstacles) compared to unrealistic scenario (i.e. scenario without obstacles) due to wireless signal attenuation and absorption.

This study proposes 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, the nodes that know their locations are called reference nodes, and the other nodes that are without the knowledge of their locations are called blind nodes. To localize a certain blind node, by comparing the relative RSSI (Received Signal Strength Indicator) values among nodes, mobile beacons are utilized to find out the rings that are centered at a reference node and contain the blind node. These rings are called B-Rings. Since the mobile anchors and the reference nodes know their own locations, the B-Rings can be precisely derived. Moreover, by using multiple 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. Most existing localization schemes that utilize mobile anchors let the mobile anchors move randomly. In contrast, the proposed scheme provides regular and simple movement mechanisms for the mobile anchors. Thus, the mobile anchors consume less energy than the other schemes, in which the mobile anchors move randomly. Analytical analysis and simulation results show that the proposed localization mechanism can achieve better location accuracy as well as less movement length of the mobile anchor than the other existing related approaches.

In performing video multicast over WiMAX networks, AMC enables the BSs to be able to select an appropriate modulation method in accordance with the transmission objective, namely to improve the video quality or to maximize the bandwidth utilization efficiency. However, by choosing the modulation method which maximizes the video quality at all of the SSs, the bandwidth utilization efficiency is reduced and vice versa. In addition, because wireless channels are noisy, FEC is always employed for video multicast. However, by choosing high FEC ratio, the video quality is improved but the bandwidth utilization efficiency is degraded and vice versa. In other words, a tradeoff exists between the video quality and the bandwidth utilization efficiency for video multicast over WiMAX networks. Accordingly, the present study proposes a scheme designated as Performance-Driven Robust Video Multicast PDRVM, in which both the modulation method and the FEC ratio are determined adaptively in accordance with the network status by mathematical analyses. Simulation results show that PDRVM not only ensures that most of the SSs acquire a good video quality, but also makes the best possible use of the available transmission resources.

According to previous work, the performance of the Distributed Coordination Function (DCF) (i.e., the basic access method of the IEEE 802.11 protocol) is far from optimum due to use of the binary exponential backoff (BEB) scheme as its collision avoidance mechanism. There has been considerable discussion of DCF issues and its performance analysis. However, most schemes assume an ideal channel, which is contrary to realistic wireless environments. In this paper, we present a simple yet pragmatic distributed algorithm, designated the density based access method (DBM), which allows stations to dynamically optimize the network throughput based on run-time measurements of the channel status. Our simulation results demonstrate that the DBM is highly accurate. The performance in terms of throughput and fairness is nearly optimal by use of the proposed scheme.

Traditional Forward Error Correction (FEC) mechanisms can be divided into Packet level FEC (PFEC) mechanisms and Byte level FEC (BFEC) mechanisms. The PFEC mechanism of recovering from errors in a source packet requires an entire FEC redundant packet even though the error involves a few bit errors. The recovery capability of the BFEC mechanism is only half of the FEC redundancy. Accordingly, an adaptive Sub-Packet FEC (SPFEC) mechanism is proposed in this paper to improve the quality of video streaming data over wireless networks, simultaneously enhancing the recovery performance and reducing the end-to-end delay jitter. The SPFEC mechanism divides a packet into n sub-packets by means of the concept of a virtual packet. The SPFEC mechanism uses a checksum in each sub-packet to identify the position of the error sub-packet. Simulation experiments show the adaptive SPFEC mechanism achieves high recovery performance and low end-to-end delay jitter. The SPFEC mechanism outperforms traditional FEC mechanism in terms of packet loss rate and video Peak Signal-to-Noise Ratio (PSNR). SPFEC offers an alternative for improved efficiency video streaming that will be of interest to the designers of the next generation environments.

According to the latest version of the IEEE 802.11 standard, the backoff parameters of its collision avoidance mechanism are far from optimal, especially in a heavy load or error-prone WLAN environment. This strategy has a high collision probability and channel utilization is degraded in bursty arrivals or congested scenarios. Besides, the standard backoff mechanism may treat noise corruption as packet collisions. In this paper, we identify the relationship between backoff parameters, contention level, and channel BER in order to propose a simple, but yet well-performing distributed algorithm that allows a station to dynamically adjust its contention window size based on turn-around-time measurement of channel status. In addition to theoretical analysis, simulations are conducted to evaluate its performance. The proposed scheme works very well in providing a substantial performance improvement in heavy loaded and error-prone WLAN environments.

This paper proposes a Random Early Detection Forward Error Correction (RED-FEC) mechanism to improve the quality of video delivered over Wireless Local Area Networks (WLANs). In contrast to previous FEC schemes, in which the rate determination information is fed back from the receiver side, in our proposed method, the redundancy rate is calculated directly at the wireless Access Point (AP) in accordance with the network traffic load, as indicated by the AP queue length. An analytical model is developed to predict the effective packet loss rate of a video stream delivered over a WLAN with RED-FEC protection. The numerical results show that the proposed RED-FEC mechanism consistently achieves higher recovery efficiency than a conventional FEC scheme under high and low network loading conditions.

The failure of legacy packet markers, such as the Single Rate Three Color Marker (SRTCM) and the Two Rate Three Color Marker (TRTCM), to protect important data from being dropped, renders a poor delivery of video quality over Differentiated Services (DiffServ) networks due to the hierarchical structure of MPEG video. Therefore, we propose a content-aware packet marking scheme, which we call Two Markers System (TMS), to improve the delivery quality of MPEG video streams. TMS places marker modules at both the video source and the edge of a DiffServ network. The source marking modules act at the application-level to set an importance indicator in the IP header of the MPEG frame packet by evaluating the contribution of the packet to the expected picture quality perceived by an end user. Meanwhile, at the edge of the DiffServ network, the Enhanced Token Bucket Three Color Marker (ETBTCM) is proposed to perform packet marking at the network-level in accordance with the pre-assigned importance of individual video frame packets. When the traffic flow is measured as out-of-profile, the less important packets will be marked with higher probability than the more important ones. Therefore, in the event of network congestion, less important frame packets can be dropped before more important ones. Consequently, a better delivered video quality can be provided. Further, due to the convenience offered by wireless networks, more users choose mobile devices to connect to the Internet. Therefore, a Video Differentiator is suggested, placed at the entry to the IEEE 802.11e wireless network, to provide traffic differentiation and better service for video traffic. We use authentic MPEG4 video traffic traces to evaluate this framework. Results show that better video quality can be delivered over our proposed new framework.

The main purpose of this article is to provide researchers or practitioners with greater flexibility to analyze their proposed designs for network, protocol, or video coding of MDC video transmission without being obliged to consider a particular tool set for each kind of video-quality evaluation. Therefore, a generalized simulation and evaluation frame work for video transmission is an important and useful tool to the video-coding of networking was found by research community. On the one hand, for video-coding studies, one can simulate the effects of a more realistic network on video sequences resulting from their coding schemes. On the other hand, in networking analysis, one can evaluate the impact of real video streams on proposed network protocols

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.

Recent developments in hardware, software and communication technologies have resulted in increasing interest in the use of wireless local area networks (WLANs). Mobile devices with embedded WLAN functionality are becoming increasingly popular. Such devices must be designed to support applications that require high quality of service (QoS) and have favorable to maximize battery capacity. The resources of queues in IEEE 802.11e networks may be wasted by the transmission of information that is useless to the receiver. This work develops a frame-based mapping mechanism (FBM) that exploits different methods to process I/P/B (Intra/Predictive/Bipredictive) video frame packets. FBM refers to the dropping of arriving packets if the preceding packets in the same video frame have been dropped. When fragmented packets of a single frame are allocated to different access categories (AC) queues, out-of order delivery may occur. Hence, FBM tries to treat all fragmented packets of each video frame equally and allocates them to the same AC queue if possible. The simulation results demonstrate that transmission by the FBM is more efficient than that by other mechanisms, such as EDCA (Enhanced Distributed Channel Access), static mapping and adaptive mapping, suggesting that the energy of a device is not wasted in the transmission of useless video data in WLANs. (C) 2015 Elsevier Ltd. All rights reserved., Foundation item: The National Project of Taiwan (No.: MOST 103-2221-E507-001). Authors are grateful to Ministry of Science and Technology Grant no. (MOST 103-2221-E507-001), Government of Taiwan for financial support to carry out this work.

Recently, voice over Internet Protocol (VoIP) has gained much attention in wireless local area networks (WLANs). Although IEEE 802.11e enhanced distributed channel access (EDCA) has been standardized to provide voice traffic with quality of service (QoS) support, the voice quality could still be severely degraded by the traffic transmitted from low-priority stations (LP-STAs). Like 802.11b, the 802.11e EDCA also has the performance anomaly problem. Therefore, the voice quality of the high-rate stations (HR-STAs) could be severely degraded by the traffic transmitted from low-rate stations (LR-STAs). In fact, the LP-STAs and LR-STAs can be present simultaneously in practical WLAN environments. In this paper, we consider the effects of LP-STAs and/or LR-STAs on voice traffic and propose an extended EDCA scheme called voice differentiation-EDCA (VD-EDCA) to protect voice traffic from LP-STAs and/or LR-STAs. With VD-EDCA, the voice AC in a HR-STA has an absolute priority over LP-STAs and LR-STAs when they are notified that the wireless delay of a voice frame at the QoS access point (QAP) is larger than or equal to the predefined maximal delay threshold. We evaluate the performance of the proposed VD-EDCA scheme through extensive simulations. The simulation results show that, with appropriate maximal delay threshold settings, the proposed scheme well protects all of the VoIP calls transmitted by HR-STAs from LP-STAs and/or LR-STAs

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.

As wireless local area networks have been widely deployed and smart devices are becoming common in human everyday life, the demanding for video applications, such as video conference, video on demand, and video games is still increasing. However, the bandwidth in wireless networks is limited and channel quality is varying. As a result, how to provide a better delivered video quality over wireless networks is challenging. Also, the current solutions to better video transmission are almost only considering single hop transmission. When those solutions are applied in multi-hop wireless networks, such as wireless mesh network or VANET, the delivered video quality may not be good as one hop transmission. Therefore, we propose a new mechanism to improve video transmission in multi-hop wireless networks. The new mechanism contains two methods. One is to privilege the packets that have already traversed more hops. The other is control the number of retransmission for video packets with different importance. Through NS2 simulations, the results show the effectiveness of proposed mechanism.

The failure of legacy packet markers such as the Single Rate Three Color Marker (SRTCM) and the Two Rate Three Color Marker (TRTCM) to distinguish important and less important data renders them unsuitable for MPEG video transmission in a DiffServ network. Therefore, we propose a Two Markers System (TMS) to improve the delivery quality of MPEG video streams. We use authentic MPEG4 video traffic traces to compare the performance of the proposed TMS system with those of legacy packet markers. Results show that TMS outperforms the legacy traffic markers in terms of the quality of the delivered MPEG video streams.

This study proposed a High performance Distributed Coordination Function with QoS support (QHDCF) protocol to enhance the throughput and achieve fair allocation of wireless medium access. QHDCF protocol has two working modes: contending mode and active mode. In the contending mode, all stations follow DCF scheme to contend for channel access. Once a station successfully accesses the channel, it switches to the active mode. In the active mode, the transmitting station determines the next transmitter according to a selection scheme from a probability formulation perspective. The selection scheme is based on relative weights of different traffic classes so that weighted-fairness can be achieved. Besides, since most stations do not contend for channel access, collisions are reduced a lot and therefore the throughput is enhanced. Simulation results show that QHDCF performs significantly better than EDCA in terms of both weighted fairness and throughput.

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 ad-hoc network (VANET) is a rapidly developing technology which supports vehicle-to-vehicle communications and vehicle-to-infrastructure communications. With the two types of communication modes, users can access the Internet via some devices such as CAR PCs and PDAs. With the Internet connection, many value-added applications can be developed. However, Internet connection for VANETs faces a great challenge: the high mobility of vehicles results in frequent handoffs, which may cause significant packet delay and packet losses. This paper examines the necessary steps involved in a VANET handoff process and reviews some related studies in the literature that reduces the handoff time for VANETs.

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.

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.

According to the latest version of the IEEE 802.16 standard, the mandatory contention resolution method is the truncated binary exponential backoff, with the initial window size and the maximum window size controlled by the base station. However, the problem of choosing the right set of backoff parameters for the current network level remains unsolved and left as an open issue since this strategy might incur a high collision probability and the channel utilization could be degraded in congested scenario. In this paper, we propose two pragmatic adaptive algorithms, namely semi-dynamic and quasi-dynamic contention resolution schemes, that allow the base station to adjust its backoff window size based on current channel status. By controlling the size of backoff window according to varying network conditions, both schemes are able to achieve higher performance in comparison with the legacy IEEE 802.16 standard.

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.

The QoS solutions for video transmission over IP network presently include integrated services (IntServ) [4], DiffServ [5], multiprotocol label switching (MPLS) [6]. Among them, the DiffServ is regarded as a dominant protocol for its flexibility, scalability, and capability of QoS guarantee. Traditionally, policing or limiting function was implemented in DiffServ by using srTCM (single-rate three color marker) [2] and trTCM (two-rate three-color marker) [3]. Proposed scheme improve the throughput of MPEG4 video transmissions over IP network. In this scheme, I-frames, P-frames and B-frames are marked as different service priorities according to transmission condition and their contribution to the perceived picture quality. Investigators used authentic MPEG4 video traffic traces to compare the performance of the proposed scheme with other exesting RFCs for MPEG video transmission, such as srTCM and trTCM, in a DiffServ network. Results show that proposed scheme improves the throughput of video transmission over IP network.

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.