76 research outputs found
IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH
Voice over Internet Protocol (VoIP) is a technology that allows the transmission of
voice packets over Internet Protocol (IP). Recently, the integration of VoIP and
Wireless Local Area Network (WLAN), and known as Voice over WLAN
(VoWLAN), has become popular driven by the mobility requirements ofusers, as
well as by factor of its tangible cost effectiveness. However, WLAN network
architecture was primarily designed to support the transmission of data, and not for
voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS)
for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11
standards that support Link Adaptive (LA) technique. However, LA leads to having a
network with multi-rate transmissions that causes network bandwidth variation, which
hence degrades the voice quality. Therefore, it is important to develop an algorithm
that would be able to overcome the negative effect of the multi-rate issue on VoIP
quality. Hence, the main goal ofthis research work is to develop an agent that utilizes
IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned
negative effect. This could be expected from the interaction between Medium Access
Control (MAC) layer and Application layer, where the proposed agent adapts the
voice packet size at the Application layer according to the change of MAC
transmission data rate to avoid network congestion from happening. The agent also
monitors the quality of conversations from the periodically generated Real Time
Control Protocol (RTCP) reports. If voice quality degradation is detected, then the
agent performs further rate adaptation to improve the quality. The agent performance
has been evaluated by carrying out an extensive series ofsimulation using OPNET
Modeler. The obtained results of different performance parameters are presented,
comparing the performance ofVoWLAN that used the proposed agent to that ofthe
standard network without agent. The results ofall measured quality parameters hav
Recommended from our members
Towards the Quality of Service for VoIP traffic in IEEE 802.11 Wireless Networks
The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic
VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS
âVoice over Internet Protocol (VoIP)â is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in âWireless Local Area Networks (WLAN)â.
IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic
IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH
Voice over Internet Protocol (VoIP) is a technology that allows the transmission of
voice packets over Internet Protocol (IP). Recently, the integration of VoIP and
Wireless Local Area Network (WLAN), and known as Voice over WLAN
(VoWLAN), has become popular driven by the mobility requirements ofusers, as
well as by factor of its tangible cost effectiveness. However, WLAN network
architecture was primarily designed to support the transmission of data, and not for
voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS)
for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11
standards that support Link Adaptive (LA) technique. However, LA leads to having a
network with multi-rate transmissions that causes network bandwidth variation, which
hence degrades the voice quality. Therefore, it is important to develop an algorithm
that would be able to overcome the negative effect of the multi-rate issue on VoIP
quality. Hence, the main goal ofthis research work is to develop an agent that utilizes
IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned
negative effect. This could be expected from the interaction between Medium Access
Control (MAC) layer and Application layer, where the proposed agent adapts the
voice packet size at the Application layer according to the change of MAC
transmission data rate to avoid network congestion from happening. The agent also
monitors the quality of conversations from the periodically generated Real Time
Control Protocol (RTCP) reports. If voice quality degradation is detected, then the
agent performs further rate adaptation to improve the quality. The agent performance
has been evaluated by carrying out an extensive series ofsimulation using OPNET
Modeler. The obtained results of different performance parameters are presented,
comparing the performance ofVoWLAN that used the proposed agent to that ofthe
standard network without agent. The results ofall measured quality parameters hav
VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS
âVoice over Internet Protocol (VoIP)â is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in âWireless Local Area Networks (WLAN)â.
IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic
Quality of service differentiation for multimedia delivery in wireless LANs
Delivering multimedia content to heterogeneous devices over a variable networking environment while maintaining high quality levels involves many technical challenges. The research reported in this thesis presents a solution for Quality of Service (QoS)-based service differentiation when delivering multimedia content over the wireless LANs. This thesis has three major contributions outlined below:
1. A Model-based Bandwidth Estimation algorithm (MBE), which estimates the available bandwidth based on novel TCP and UDP throughput models over IEEE 802.11 WLANs. MBE has been modelled, implemented, and tested through simulations and real life testing. In comparison with other bandwidth estimation techniques, MBE shows better performance in terms of error rate, overhead, and loss.
2. An intelligent Prioritized Adaptive Scheme (iPAS), which provides QoS service differentiation for multimedia delivery in wireless networks. iPAS assigns dynamic priorities to various streams and determines their bandwidth share by employing a probabilistic approach-which makes use of stereotypes. The total bandwidth to be allocated is estimated using MBE. The priority level of individual stream is variable and dependent on stream-related characteristics and delivery QoS parameters. iPAS can be deployed seamlessly over the original IEEE 802.11 protocols and can be included in the IEEE 802.21 framework in order to optimize the control signal communication. iPAS has been modelled, implemented, and evaluated via simulations. The results demonstrate that iPAS achieves better performance than the equal channel access mechanism over IEEE 802.11 DCF and a service differentiation scheme on top of IEEE 802.11e EDCA, in terms of fairness, throughput, delay, loss, and estimated PSNR. Additionally, both objective and subjective video quality assessment have been performed using a prototype system.
3. A QoS-based Downlink/Uplink Fairness Scheme, which uses the stereotypes-based structure to balance the QoS parameters (i.e. throughput, delay, and loss) between downlink and uplink VoIP traffic. The proposed scheme has been modelled and tested through simulations. The results show that, in comparison with other downlink/uplink fairness-oriented solutions, the proposed scheme performs better in terms of VoIP capacity and fairness level between downlink and uplink traffic
Contribution to quality of user experience provision over wireless networks
The widespread expansion of wireless networks has brought new attractive possibilities to end users. In addition to the mobility capabilities provided by unwired devices, it is worth remarking the easy configuration process that a user has to follow to gain connectivity through a wireless network. Furthermore, the increasing bandwidth provided by the IEEE 802.11 family has made possible accessing to high-demanding services such as multimedia communications. Multimedia traffic has unique characteristics that make it greatly vulnerable against network impairments, such as packet losses, delay, or jitter. Voice over IP (VoIP) communications, video-conference, video-streaming, etc., are examples of these high-demanding services that need to meet very strict requirements in order to be served with acceptable levels of quality. Accomplishing these tough requirements will become extremely important during the next years, taking into account that consumer video traffic will be the predominant traffic in the Internet during the next years. In wired systems, these requirements are achieved by using Quality of Service (QoS) techniques, such as Differentiated Services (DiffServ), traffic engineering, etc. However, employing these methodologies in wireless networks is not that simple as many other factors impact on the quality of the provided service, e.g., fading, interferences, etc. Focusing on the IEEE 802.11g standard, which is the most extended technology for Wireless Local Area Networks (WLANs), it defines two different architecture schemes. On one hand, the infrastructure mode consists of a central point, which manages the network, assuming network controlling tasks such as IP assignment, routing, accessing security, etc. The rest of the nodes composing the network act as hosts, i.e., they send and receive traffic through the central point. On the other hand, the IEEE 802.11 ad-hoc configuration mode is less extended than the infrastructure one. Under this scheme, there is not a central point in the network, but all the nodes composing the network assume both host and router roles, which permits the quick deployment of a network without a pre-existent infrastructure. This type of networks, so called Mobile Ad-hoc NETworks (MANETs), presents interesting characteristics for situations when the fast deployment of a communication system is needed, e.g., tactics networks, disaster events, or temporary networks. The benefits provided by MANETs are varied, including high mobility possibilities provided to the nodes, network coverage extension, or network reliability avoiding single points of failure. The dynamic nature of these networks makes the nodes to react to topology changes as fast as possible. Moreover, as aforementioned, the transmission of multimedia traffic entails real-time constraints, necessary to provide these services with acceptable levels of quality. For those reasons, efficient routing protocols are needed, capable of providing enough reliability to the network and with the minimum impact to the quality of the service flowing through the nodes. Regarding quality measurements, the current trend is estimating what the end user actually perceives when consuming the service. This paradigm is called Quality of user Experience (QoE) and differs from the traditional Quality of Service (QoS) approach in the human perspective given to quality estimations. In order to measure the subjective opinion that a user has about a given service, different approaches can be taken. The most accurate methodology is performing subjective tests in which a panel of human testers rates the quality of the service under evaluation. This approach returns a quality score, so-called Mean Opinion Score (MOS), for the considered service in a scale 1 - 5. This methodology presents several drawbacks such as its high expenses and the impossibility of performing tests at real time. For those reasons, several mathematical models have been presented in order to provide an estimation of the QoE (MOS) reached by different multimedia services In this thesis, the focus is on evaluating and understanding the multimedia-content transmission-process in wireless networks from a QoE perspective. To this end, firstly, the QoE paradigm is explored aiming at understanding how to evaluate the quality of a given multimedia service. Then, the influence of the impairments introduced by the wireless transmission channel on the multimedia communications is analyzed. Besides, the functioning of different WLAN schemes in order to test their suitability to support highly demanding traffic such as the multimedia transmission is evaluated. Finally, as the main contribution of this thesis, new mechanisms or strategies to improve the quality of multimedia services distributed over IEEE 802.11 networks are presented. Concretely, the distribution of multimedia services over ad-hoc networks is deeply studied. Thus, a novel opportunistic routing protocol, so-called JOKER (auto-adJustable Opportunistic acK/timEr-based Routing) is presented. This proposal permits better support to multimedia services while reducing the energy consumption in comparison with the standard ad-hoc routing protocols.Universidad PolitĂ©cnica de CartagenaPrograma Oficial de Doctorado en TecnologĂas de la InformaciĂłn y Comunicacione
Quality of Service Provisioning for Voice Application over Wlans
RĂ©sumĂ© Aujourd'hui, les rĂ©seaux locaux sans fils WLANs (en anglais Wireless Local Area Networks) sont de plus en plus dĂ©ployĂ©s en raison de leur facilitĂ© d'installation et de leur faible coĂ»ts. Le standard IEEE (en anglais Institute of Electrical and Electronics Engineers) 802.11 dĂ©finit les caractĂ©ristiques des rĂ©seaux locaux sans fil WLANs. Alors que la premiĂšre norme proposĂ©e pouvait soutenir un dĂ©bit de donnĂ©es jusqu'Ă 2 Mbps, dans les versions rĂ©centes de la norme 802.11, des dĂ©bits de donnĂ©es pouvant atteindre jusqu'Ă 200 Mbps seront pris en charge dans les rĂ©seaux sans fil de prochaine gĂ©nĂ©ration. Garantir les besoins de QoS (en anglais Quality of Service) est un dĂ©fi considĂ©rable pour les rĂ©seaux WLAN, en particulier pour les applications multimĂ©dia. Vu que largeur de bande disponible dans les rĂ©seaux sans fil est limitĂ©e, la bande passante supportant la QoS ne peut ĂȘtre facilement augmentĂ©e. Cependant, les protocoles efficaces capables de satisfaire la qualitĂ© de service doivent ĂȘtre conçus pour amĂ©liorer l'utilisation des ressources dans les rĂ©seaux. Le protocole de la couche MAC (en anglais Medium Access Control) affecte fondamentalement les paramĂštres QoS. Le contrĂŽle d'admission est Ă©galement un Ă©lĂ©ment essentiel pour la qualitĂ© de service dans les rĂ©seaux locaux sans fil.
Ca projet a pour objectif de modéliser et analyser la couche MAC et contrÎler l'admission dans des réseaux locaux sans fil 802.11 avec infrastructure basée sur le mécanisme DCF (en anglais Distributed Coordination Function). Bien que notre objectif général est de garantir les besoins de QoS des applications multimédias sur les réseaux sans fil, nous avons répondu à plusieurs questions importantes telles que la modélisation de la couche MAC, l'évaluation de la QoS et le contrÎle d'admission.
La premiĂšre contribution de cette recherche est de proposer un cadre analytique qui prend en compte la direction du trafic en mode infrastructure non saturĂ©. Contrairement aux recherches antĂ©rieures, dans l'analyse proposĂ©e la probabilitĂ© de collision d'un paquet transmis par chaque station sans fil en liaison ascendante est diffĂ©rente de la probabilitĂ© de collision pour les paquets qui sont transmis Ă partir du point d'accĂšs en liaison descendante. Ce modĂšle de distinction entre les modĂšles backoff par station en liaison ascendante et en liaison descendante est capable d'exprimer la performance MAC en termes de nombre de stations sans fil et plusieurs paramĂštres systĂšme tels que la taille de la fenĂȘtre de contention, le nombre maximum d'Ă©tapes backoff, la taille du tampon Ă la couche MAC, ainsi que les paramĂštres de trafic tels que les durĂ©es de conversation, le silence et le taux d'arrivĂ©e. Contrairement aux Ă©tudes prĂ©cĂ©dentes, nous appliquons deux groupes d'Ă©quations, un groupe est dĂ©fini pour la station sans fil et l'autre pour le point d'accĂšs. ----------Abstract Wireless Local Area Networks (WLANs) are widely deployed nowadays because of their low cost and convenient implementation. The Institute of Electrical and Electronics Engineers (IEEE) 802.11 series standards define the specifications for such networks. While the first proposed standard could support a data rate up to 2 Mbps, in the recent upgraded versions of the standard data rates up to 54 Mbps are achievable and up to 200 Mbps are said to be supported in next generation WLANs. An important concern in WLANs is the support of Quality of Service (QoS), specifically for multimedia applications. Because of the limited available bandwidth in wireless networks, bandwidth cannot be easily increased to support QoS. However, efficient protocols capable of providing QoS have to be designed to improve resource utilization in networks. The Medium Access Control (MAC) protocol crucially affects the QoS parameters. Admission control is also an essential element for QoS provisioning in WLANs.
Our research covers mathematical modeling and analysis of the MAC layer and admission control considering Distributed Coordination Function (DCF) in infrastructure mode of IEEE 802.11-based WLANs. While our general goal is to guarantee the QoS parameters of multimedia applications over WLANs, we address several important issues such as MAC layer modeling, QoS evaluation and admission control.
The first contribution of this research is to propose an analytical framework which takes into account the traffic direction in non-saturated infrastructure mode of WLANs. Unlike previous work, in the proposed analysis the collision probability of a packet transmitted by each wireless station in the uplink direction is different from the probability of collision for the packets transmitted from the access point in the downlink direction. Our model differentiates between per-station backoff models in the uplink and downlink and is capable of expressing the MAC performance in terms of several system parameters such as contention window size, maximum number of backoff stages, size of buffer at the MAC layer, traffic parameters such as talk and silent durations and arrival rate, as well as the number of wireless stations. In contrast to the previous studies, we apply two groups of equations, one group is defined for the wireless station and the other one for the access point. These equations represent the transmission probability, probability of collision and the probability of being in the busy state in terms of the number of wireless stations, the traffic arrival rate and system parameters such as the size of the contention window and maximum number of retransmissions
Recommended from our members
Design of interface selection protocols for multi-homed wireless networks
This thesis was submitted for the degree of Doctor of Philosophy and was awarded by Brunel University on 10 December 2010.The IEEE 802.11/802.16 standards conformant wireless communication stations have multi-homing transmission capability. To achieve greater communication efficiency, multi-homing capable stations use handover mechanism to select appropriate transmission channel according to variations in the channel quality. This thesis presents three internal-linked handover schemes, (1) Interface Selection Protocol (ISP), belonging to Wireless Local Area Network (WLAN)- Worldwide Interoperability for Microwave Access (WiMAX) environment (2) Fast Channel Scanning (FCS) and (3) Traffic Manager (TM), (2) and (3) belonging to WiMAX Environment. The proposed schemes in this thesis use a novel mechanism of providing a reliable communication route. This solution is based on a cross-layer communication framework, where the interface selection module uses various network related parameters from Medium Access Control (MAC) sub-layer/Physical Layer (PHY) across the protocol suite for decision making at the Network layer. The proposed solutions are highly responsive when compared with existing multi-homed schemes; responsiveness is one of the key factors in the design of such protocols. Selected route under these schemes is based on the most up to date link-layer information. Therefore, such a route is not only reliable in terms of route optimization but it also fulfils the application demands in terms of throughput and delay. Design of ISP protocol use probing frames during the route discovery process. The 802.11 mandates the use of different rates for data transmission frames. The ISP-metric can be incorporated into various routing aspects and its applicability is determined by the possibility of provision of MAC dependent parameters that are used to determine the best path metric values. In many cases, higher device density, interference and mobility cause variable medium access delays. It causes creation of âunreachable zonesâ, where destination is marked as unreachable. However, by use of the best path metric, the destination has been made reachable, anytime and anywhere, because of the intelligent use of the probing frames and interface selection algorithm implemented. The IEEE 802.16e introduces several MAC level queues for different access categories, maintaining service requirement within these queues; which imply that frames from a higher priority queue, i.e. video frames, are serviced more frequently than those belonging to lower priority queues. Such an enhancement at the MAC sub-layer introduces uneven queuing delays. Conventional routing protocols are unaware of such MAC specific constraints and as a result, these factors are not considered which result in channel performance degradation. To meet such challenges, the thesis presents FCS and TM schemes for WiMAX. For FCS, Its solution is to improve the mobile WiMAX handover and address the scanning latency. Since minimum scanning time is the most important issue in the handover process. This handover scheme aims to utilize the channel efficiently and apply such a procedure to reduce the time it takes to scan the neighboring access stations. TM uses MAC and physical layer (PHY) specific information in the interface metric and maintains a separate path to destination by applying an alternative interface operation. Simulation tests and comparisons with existing multi-homed protocols and handover schemes demonstrate the effectiveness of incorporating the medium dependent parameters. Moreover, show that suggested schemes, have shown better performance in terms of end-to-end delay and throughput, with efficiency up to 40% in specific test scenarios
A Unified Mobility Management Architecture for Interworked Heterogeneous Mobile Networks
The buzzword of this decade has been convergence: the convergence of telecommunications, Internet, entertainment, and information technologies for the seamless provisioning of multimedia services across different network types. Thus the future Next Generation Mobile Network (NGMN) can be envisioned as a group of co-existing heterogeneous mobile data networking technologies sharing a common Internet Protocol (IP) based backbone. In such all-IP based heterogeneous networking environments, ongoing sessions from roaming users are subjected to frequent vertical handoffs across network boundaries. Therefore, ensuring uninterrupted service continuity during session handoffs requires successful mobility and session management mechanisms to be implemented in these participating access networks. Therefore, it is essential for a common interworking framework to be in place for ensuring seamless service continuity over dissimilar networks to enable a potential user to freely roam from one network to another. For the best of our knowledge, the need for a suitable unified mobility and session management framework for the NGMN has not been successfully addressed as yet. This can be seen as the primary motivation of this research. Therefore, the key objectives of this thesis can be stated as: To propose a mobility-aware novel architecture for interworking between heterogeneous mobile data networks To propose a framework for facilitating unified real-time session management (inclusive of session establishment and seamless session handoff) across these different networks. In order to achieve the above goals, an interworking architecture is designed by incorporating the IP Multimedia Subsystem (IMS) as the coupling mediator between dissipate mobile data networking technologies. Subsequently, two different mobility management frameworks are proposed and implemented over the initial interworking architectural design. The first mobility management framework is fully handled by the IMS at the Application Layer. This framework is primarily dependant on the IMSâs default session management protocol, which is the Session Initiation Protocol (SIP). The second framework is a combined method based on SIP and the Mobile IP (MIP) protocols, which is essentially operated at the Network Layer. An analytical model is derived for evaluating the proposed scheme for analyzing the network Quality of Service (QoS) metrics and measures involved in session mobility management for the proposed mobility management frameworks. More precisely, these analyzed QoS metrics include vertical handoff delay, transient packet loss, jitter, and signaling overhead/cost. The results of the QoS analysis indicates that a MIP-SIP based mobility management framework performs better than its predecessor, the Pure-SIP based mobility management method. Also, the analysis results indicate that the QoS performances for the investigated parameters are within acceptable levels for real-time VoIP conversations. An OPNET based simulation platform is also used for modeling the proposed mobility management frameworks. All simulated scenarios prove to be capable of performing successful VoIP session handoffs between dissimilar networks whilst maintaining acceptable QoS levels. Lastly, based on the findings, the contributions made by this thesis can be summarized as: The development of a novel framework for interworked heterogeneous mobile data networks in a NGMN environment. The final design conveniently enables 3G cellular technologies (such as the Universal Mobile Telecommunications Systems (UMTS) or Code Division Multiple Access 2000 (CDMA2000) type systems), Wireless Local Area Networking (WLAN) technologies, and Wireless Metropolitan Area Networking (WMAN) technologies (e.g., Broadband Wireless Access (BWA) systems such as WiMAX) to interwork under a common signaling platform. The introduction of a novel unified/centralized mobility and session management platform by exploiting the IMS as a universal coupling mediator for real-time session negotiation and management. This enables a roaming user to seamlessly handoff sessions between different heterogeneous networks. As secondary outcomes of this thesis, an analytical framework and an OPNET simulation framework are developed for analyzing vertical handoff performance. This OPNET simulation platform is suitable for commercial use
- âŠ