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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.

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.

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.

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.

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.

In this paper, we proposed a backoff mechanism to improve the performance of initial ranging for the IEEE 802.16 standard. In the proposed scheme, the size of contention window in the initial ranging is combined with exponential and linear backoff mechanisms, and it is adjusted based on the current network environments. In addition, a threshold in the proposed scheme is set to find out the current network situation. If the contention window size is smaller than the threshold, the contention window size is tuned exponentially in response to a light network traffic load. On the other hand, when the contention window size is larger than the threshold, the contention window size is tuned linearly in response to a congested scenario. The numerical simulation results show that the proposed scheme can achieve a better performance in terms of collision rate, average access delay, and system throughput in both light and congested environments.

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.

In this paper, a backoff mechanism, Exponential Linear Backoff Algorithm (ELBA), is proposed to improve system performance over contention-based wireless networks. In the ELBA, the variation of contention window size is combined both exponentially and linearly, dependent on the network load, as indicated by the number of consecutive collisions. In the ELBA scheme, a threshold is set to determine the network load. If the contention window size is smaller than the threshold, a light network load, the contention window is tuned exponentially. Conversely, if the contention window size is larger than the threshold, a heavy network load, the contention window size is tuned linearly. The numerical results show that the ELBA provides a better system throughput and collision rate in both light and heavy network loads than the related backoff schemes, including binary exponential backoff (BEB), exponential increase exponential decrease (EIED) and linear increase linear decrease (LILD).

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.

Presenting an advanced simulation tool-set for video transmission performance evaluation is the objective of this paper. This tool-set integrates NS-2 with Evalvid plus a new coding mechanism: Multiple Description Coding (MDC). Using the integrated NS-2 simulator, researchers can easily evaluate specific transmission performance of a variety of videos varying in codec or coding mechanisms. Researchers can analyze their own design, such as their own improved network protocols or QoS control schemes, in a realistic simulation environment without delving into the simulator itself. Besides, by using the tool-set, researchers can acquire video transmission quality not only with former evaluation metrics, but also with real recombined video, which can be played out and observed. In conclusion, researchers who utilize our evolved simulation tool-set will benefit from speedily verifying their video transmission over wireless network designs.

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

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.

SUMMARY Unequal error protection systems are a popular technique for video streaming. Forward error correction (FEC) is one of error control techniques to improve the quality of video streaming over lossy channels. Moreover, frame-level FEC techniques have been proposed for video streaming because of different priority video frames within the transmission rate constraint on a Bernoulli channel. However, various communication and storage systems are likely corrupted by bursts of noise in the current wireless behavior. If the burst losses go beyond the protection capacity of FEC, the efficacy of FEC can be degraded. Therefore, our proposed model allows an assessment of the perceived quality of H.264/AVC video streaming over bursty channels, and is validated by simulation experiments on the NS-2 network simulator at a given estimate of the packet loss ratio and average burst length. The results suggest a useful reference in designing the FEC scheme for video applications, and as the video coding and channel parameters are given, the proposed model can provide a more accurate evaluation tool for video streaming over bursty channels and help to evaluate the impact of FEC performance on different burst-loss parameters. Copyright © 2014 John Wiley & Sons, Ltd.