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  • About
  • The Global ETD Search service is a free service for researchers to find electronic theses and dissertations. This service is provided by the Networked Digital Library of Theses and Dissertations.
    Our metadata is collected from universities around the world. If you manage a university/consortium/country archive and want to be added, details can be found on the NDLTD website.
1

Uplink Scheduling for Supporting Packet Voice Traffic in IEEE 802.16 Backhaul Networks

Dai, Lizhong 09 1900 (has links)
<p> Wireless metropolitan area networking based on IEEE 802.16 is expected to be widely used for creating wide-area wireless backhaul networks, where each subscriber station (SS) is responsible for forwarding traffic for a number of connections. Quality of Service (QoS) provisioning is an important aspect in such networks. The IEEE 802.16 standard specifies that the bandwidth requests sent by the SS are for individual connections and pass only the number of bytes requested from each connection. This is inefficient for backhaul networks where each SS may be responsible for forwarding packets for a relatively large number of connections and the bandwidth request messages consume much bandwidth unnecessarily. Furthermore, the standard does not include latency information, which makes it difficult for the base station (BS) to schedule real-time traffic. </p> <p> In this thesis we study real-time voice traffic support in IEEE 802.16-based backhaul networks. We propose a simple enhancement to the bandwidth request mechanism in 802.16 for supporting packet voice traffic. First, the SS combines the bandwidth requests of multiple voice connections, which are associated to it and have the same traffic parameters, and aggregates the bandwidth requests to the BS. This makes the bandwidth request process more efficient by saving transmission time of both the BS and the SSs. Second, in order to facilitate the BS to make resource allocation decisions, the aggregate bandwidth requests include information about the latency requirements of buffered real-time packets at the SSs. We propose three different bandwidth request and packet scheduling schemes, each of which requires a different amount of information in the bandwidth requests. Our results show that the proposed bandwidth request and scheduling schemes achieve significantly lower packet loss probability than standard 802.16 bandwidth requests and weighted round robin. The results further show that there is an optimum point about how much delay information the SS should report to the BS in order to best utilize the uplink resources while providing satisfactory real-time performance for the voice traffic. </p> / Thesis / Master of Applied Science (MASc)
2

Voice and rural wireless mesh community networks: a framework to quantify scalability and manage end-user smartphone battery consumption

Om, Shree January 2021 (has links)
Philosophiae Doctor - PhD / Community wireless mesh initiatives are a pioneering option to cheap ‘last-mile’ access to network services for rural low-income regions primarily located in Sub-Saharan Africa and Developing Asia. However, researchers have criticized wireless mesh networks for their poor scalability; and scalability quantification research has mostly consisted of modularization of per-node throughput capacity behaviour. A scalability quantification model to design wireless mesh networks to provide adequate quality of service is lacking. However, scalability quantification of community mesh networks alone is inadequate because rural users need affordable devices for access; and they need to know how best to use them. Low-cost low-end smartphones offer handset affordability solutions but require smart management of their small capacity battery. Related work supports the usage of Wi-Fi for communication because it is shown to consume less battery than 2G, 3G or Bluetooth. However, a model to compare Wi-Fi battery consumption amongst different low-end smartphones is missing, as is a comparison of different over-the-top communication applications.
3

Management of low and variable bit rate ATM Adaptation Layer Type 2 traffic

Voo, Charles January 2003 (has links)
Asynchronous Transfer Mode (ATM) Adaptation Layer Type 2 (AAL2) has been developed to carry low and variable bit rate traffic. It provides high bandwidth efficiency with low packing delay by allowing voice traffic from different AAL2 channels to be multiplexed onto a single ATM virtual channel connection. Examples of where AAL2 are used include the Code Division Multiple Access and the Third Generation mobile telephony networks. The main objective of this thesis is to study traditional and novel AAL2 multiplexing methods and to characterise their performance when carrying low and variable bit rate (VBR) voice traffic. This work develops a comprehensive QoS framework which is used as a basis to study the performance of the AAL2 multiplexer system. In this QoS framework the effects of packet delay, delay variation, subjective voice quality and bandwidth utilisation are all used to determine the overall performance of the end-to-end system for the support of real time voice communications. Extensions to existing AAL2 voice multiplexers are proposed and characterised. In the case where different types of voice applications are presented to the AAL2 multiplexer, it was observed that increased efficiency gains are possible when a priority queuing scheme is introduced into the traditional AAL2 multiplexer system. Studies of the voice traffic characteristics and their effects on the performance of the AAL2 multiplexer are also investigated. It is shown that particular source behaviours can have deleterious effect on the performance of the AAL2 multiplexer. Methods of isolating these voice sources are examined and the performance of the AAL2 multiplexer re-evaluated to show the beneficial effects of a particular source isolation technique. The extent to which statistical multiplexing is possible for real time variable VBR sources is theoretically examined. These calculations highlight the difficulties in multiplexing VBR real time traffic while maintaining guaranteed delay bounds for these sources. Based on these calculations, multiplexing schemes that incorporate data transfers within the real time traffic transfer are proposed as alternatives for utilising unused bandwidth caused by the VBR nature of the voice traffic.
4

Some Investigations on QoS in the Wireline-Wireless Network Interface Zone

Tewari, Maneesh 03 1900 (has links)
In the next generation of networks we will begin to see the true convergence of voice, multimedia, and data traffic. This merging of various dedicated networks will occur both in the wired and wireless domains. Given the growth in the areas of wireless voice and data, we see that the combination of mobile and Internet communication constitutes the driving force behind the third-generation wireless system and makes the basis for the fourth-generation wireless system. For services like voice over IP over wireless (VoIPoW), the main challenge is to achieve QoS and spectrum efficiency. In order to support better QoS the IETF Mobile IP Working Group is discussing a number of enhancements to the base protocol to reduce the latency, packet loss and signaling overhead experienced during handoff. This support also includes both the call admission and the subsequent scheduling of packet transmissions. In this thesis, we will first survey the work done on issues related to QoS provisioning for wireless network and then will address bandwidth allocation problem in packet radio network with special emphasis on wireline to wireless internetworking zone. The main aim of the thesis is to evolve a strategy to reduce the call dropping probability by negotiating the QoS in those conditions when we do not have the sufficient resources (mainly bandwidth) to allocate. In order to achieve the QoS we have investigated the behaviour of the Real-time Voice traffic on a wireless link and its relation to the associated quality of service. This investigation opens a way for QoS negotiation, in a condition like during handoff, when the network is not able to sustain the negotiated bandwidth. The main results of this work are, that even with reduced bandwidth, quality for speech can be maintained at a reasonable level and this way the call dropping can be reduced. Such a scheme is useful in those conditions when we do not have the sufficient bandwidth to allocate like during a handoff of a mobile host from one cell to another. Moreover the bandwidth is a scarce resource in wireless domain so there should be an efficient call admission control policy. Many call admission control policies are proposed in the literature; here we propose a simple scheme for real-time traffic, specially speech, in a base station which increases the system throughput. In addition to above, we have also experimented with Cellular IP, one of the implementations of proposed micro-mobility architecture to provide faster handoff and seamless mobility in wired and wireless network.

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