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

An analytical model for performance evaluation of multimedia applications over EDCA in an IEEE 802.11e WLAN

We extend the modeling heuristic of (Harsha et al. 2006. In IEEE IWQoS '06, pp 178-187) to evaluate the performance of an IEEE 802.11e infrastructure network carrying packet telephone calls, streaming video sessions and TCP controlled file downloads, using Enhanced Distributed Channel Access (EDCA). We identify the time boundaries of activities on the channel (called channel slot boundaries) and derive a Markov Renewal Process of the contending nodes on these epochs. This is achieved by the use of attempt probabilities of the contending nodes as those obtained from the saturation fixed point analysis of (Ramaiyan et al. 2005. In Proceedings ACM Sigmetrics, '05. Journal version accepted for publication in IEEE TON). Regenerative analysis on this MRP yields the desired steady state performance measures. We then use the MRP model to develop an effective bandwidth approach for obtaining a bound on the size of the buffer required at the video queue of the AP, such that the streaming video packet loss probability is kept to less than 1%. The results obtained match well with simulations using the network simulator, ns-2. We find that, with the default IEEE 802.11e EDCA parameters for access categories AC 1, AC 2 and AC 3, the voice call capacity decreases if even one streaming video session and one TCP file download are initiated by some wireless station. Subsequently, reducing the voice calls increases the video downlink stream throughput by 0.38 Mbps and file download capacity by 0.14 Mbps, for every voice call (for the 11 Mbps PHY). We find that a buffer size of 75KB is sufficient to ensure that the video packet loss probability at the QAP is within 1%

Improved multi-point communication for data and voice over IEEE 802.11b

There is a growing demand for faster, improved data and voice services in rural areas without modern telecom infrastructure. A wireless network is often the only feasible solution for providing network access in this environment, due to the sparse populations and difficult natural conditions. A system solution that incorporates the Multipoint Communication System (MCS) algorithm created by TRLabs into the available IEEE 802.11b Wireless Local Area Network (WLAN) devices was proposed and studied in this thesis. It combines the advantages of both systems, that is, the MCS’ capability of integrating Voice over Internet Protocol (VoIP) and data services and the IEEE 802.11b standard, currently the most widely used in WLAN products. A system test bed was set up inside Network Simulator-2 (NS-2). The data and VoIP performance was tested. Modifications to the original MCS algorithm to improve system performance were made throughout this thesis. In a constant rate radio channel, data performance (throughput and transmission efficiency) was measured using the original MCS algorithm, which was comparable to the standard Distribution Coordination Function (DCF) operation of IEEE 802.11b when both were simulated at similar conditions. On an 802.11b platform, the Automatic Rate Fallback (ARF) feature was incorporated into the original MCS algorithm. However, when clients with different data rates were present in the same channel, all the clients involved received unacceptably low and equal data throughput, dragged down by the low rate clients. A modified MCS data polling algorithm was proposed with the capability of repeated polling, which eliminated the negative effect of low rate clients in a multi-rate channel. In addition, the original MCS algorithm was modified to be more efficient in the voice polling process. The voice performance and data throughput were tested at various conditions. However, the one-by-one polling still resulted in very low voice transmission efficiency. The time wasted became more severe with increasing relay distance and channel rate (only 8.5% in an 11 Mbps channel at 30 km). A new voice handling process similar to Time Division Multiple Access (TDMA) mode was implemented and simulated. Its voice efficiency can be kept at 25% at any setting of relay distance and channel rate. Data transmission in the same channel can also benefit from using the new voice scheme. The normalized saturation throughput could be improved by 13.5% if there were 40 voice clients involved in an 11 Mbps channel at the relay distance of 15 km, compared to the original MCS algorithm. More improvement in voice efficiency, voice capacity, and data throughput can be achieved at longer relay distance, or with more voice calls set up

Experimental analysis of WiMAX and meshed Wi-Fi quality of service

Mestrado em Engenharia Electrónica e TelecomunicaçõesA indústria das telecomunicações tem sofrido uma evolução enorme nosúltimos anos. Tanto em termos de comunicações sem fios, como em termos deligações de banda larga, assistiu-se a uma adesão massiva por parte domercado, o que se traduziu num crescimento enorme, já que a tecnologia temque estar um passo à frente da procura, de forma a suprir as carências dosconsumidores. Assim, a evolução persegue um objectivo claro: possibilidadede possuir conectividade de banda larga em qualquer lugar e instante. Nestecontexto, aparecem as tecnologias WiMAX (Worldwide Interoperability forMicrowave Access) e WI-FI em Malha como possibilidades para atingir estefim. O tema desta dissertação incide no estudo das tecnologias de WiMAX e WI-FIem Malha, mais concretamente no estudo da Qualidade de Serviço (QoS)providenciada pelas normas IEEE 802.16 e IEEE 802.11s para serviços deVoIP e VoD. Esta tese apresenta a arquitectura desenvolvida para a correcta integração deQoS para serviços em tempo real no acesso à banda larga sem fios depróxima geração. De seguida, apresenta testes efectuados com osequipamentos disponíveis de WiMAX e WI-FI em Malha, de forma a mostrar ocorrecto comportamento da atribuição extremo-a-extremo de QoS nos cenáriosescolhidos com serviços em tempo real, bem como os efeitos da mobilidade natecnologia WI-FI em Malha. ABSTRACT: The telecommunication industry has suffered a massive evolution throughoutpast years. In terms of wireless communications, as well as broadbandconnections, we’ve seen a massive adoption by the market, which conductedinto an enormous growth, since the technology must always be one step aheadof the demand, in order to be to fulfill the needs of the consumers. Therefore,the evolution pursues one clear goal: the possibility to establish a broadbandconnection anywhere and anytime. In this context, the WiMAX (WorldwideInteroperability for Microwave Access) and Meshed WI-FI technologies appearas possibilities to reach this goal. The subject of this thesis is the study of both the WiMAX and Meshed WI-FItechnologies, and more concretely the study of the QoS provided by theIEEE802.16 and IEEE 802.11s standards to VoIP and VoD services. This thesis presents the architecture developed to provide the correctintegration of QoS for real-media traffic in next generation broadband wirelessaccess. It presents tests carried out with the available WiMAX and Meshed WI-FI equipments, to show the correct behavior in the attribution of end-to-endQoS in selected scenarios with real-time services, as well as mobility effects onWI-FI Wireless Mesh technology

ボイスIPネットワークのモデリングと高性能化

Tohoku University堀口 進課

Design and analysis of LTE-WLAN wireless router with QOS preservation

Future wireless networks are envisioned to embrace a higher level of heterogeneity whereby different wireless technologies such as Long Term Evolution UMTS (LTE), Wireless Local Area Network (WLAN), WCDMA/HSPA, WiMAX, etc, not only will coexist but will also cooperate more closely. This is motivated by the fact that several complementary characteristics exist between these technologies. For example, one technology can be used as access technology while the other can be used for backhaul. To interconnect two or more wireless technologies, the usage of routing device is inevitable. In order to preserve the Quality of Service (QoS) across these technologies which come with different QoS definitions, a more comprehensive approach is required to preserve QoS across two diverse wireless technologies i.e. Enhanced Distributed Coordination Function (EDCA) for WLAN and Uplink/Downlink packet scheduling for LTE. WLAN is reasonably priced, easy to deploy and has been enjoying a wide market acceptance especially in the indoor. The LTE is expected to be the dominant 4G cellular technology. However it will take some time before LTE can attain the same level of adoption as what WLAN has achieved especially in the consumer market. The main objective of this research project is to design an access router that enables the interworking between WLAN and LTE with QoS preservation. First, the performance of both WLAN and LTE radio interfaces are investigated independently in terms of the data rates, user/system throughput, effect of multiple access and spectral efficiency. Next, different approaches and schemes which facilitate QoS preservation between WLAN and LTE over the router are investigated and evaluated in terms of different performance metrics (voice Mean Opinion Score, video delay, video traffic received, video jitter, video packet loss rate). The design and analysis of the performance are carried out through simulation as the only feasible approach to accomplish this work. OPNET Modeler is used to model the LTE-WLAN router as well as to perform the analysis. The results of this research verify the feasibility of the proposed router architecture and the interworking paradigm. The elegance of the proposed router implementation is that it does not require massive change in the existing wireless systems, LTE and WLAN to preserve the QoS. The results of the performance analysis show that it is crucial to have a QoS preservation mechanism in the router IP layer at any potential congestion point in the wireless network, to ensure that delay-sensitive and loss-sensitive applications, such as real-time video and voice, pass through unimpeded, relative to the loss-tolerant and delay-tolerant data applications. The comparison of the designed IP QoS preservation scheme namely, Priority Queuing without Block Acknowledgement (PQ noBA) shows that it can support 50% more multimedia application across the router than the other scheme

A quality of service architecture for WLAN-wired networks to enhance multimedia support

Includes abstract.Includes bibliographical references (leaves 77-84).The use of WLAN for the provision of IP multimedia services faces a number of challenges which include quality of service (QoS). Because WLAN users access multimedia services usually over a wired backbone, attention must be paid to QoS over the integrated WLAN-wired network. This research focuses on the provision of QoS to WLAN users accessing multimedia services over a wired backbone. In this thesis, the IEEE 802.11-2007 enhanced data channel access (EDCA) mechanism is used to provide prioritized QoS on the WLAN media access control (MAC) layer, while weighted round robin (WRR) queue scheduling is used to provide prioritized QoS at the IP layer. The inter-working of the EDCA scheme in the WLAN and the WRR scheduling scheme in the wired network provides end-to-end QoS on a WLAN-wired IP network. A mapping module is introduced to enable the inter-working of the EDCA and WRR mechanisms

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