Wireless networks are growing at a phenomenal rate. This growth is causing an overcrowding of the unlicensed RF spectrum, leading to increased interference between co-located devices. Existing decentralized medium access control (MAC) protocols (e.g. IEEE 802.11a/b/g standards) are poorly designed to handle interference in such dense wireless environments. This is resulting in networks with poor and unpredictable performance, especially for delay-sensitive applications such as voice and video. This dissertation presents a practical conflict-graph (CG) based approach to designing self-organizing enterprise wireless networks (or WLANs) where interference is centrally managed by the network infrastructure. The key idea is to use potential interference information (available in the CG) as an input to algorithms that optimize the parameters of the WLAN.We demonstrate this idea in three ways. First, we design a self-organizing enterprise WLAN and show how the system enhances performance over non-CG based schemes, in a high fidelity network simulator. Second, we build a practical system for conflict graph measurement that can precisely measure interference (for a given network configuration) in dense wireless environments. Finally, we demonstrate the practical benefits of the conflict graph system by using it in an optimization framework that manages associations and traffic for mobile VoIP clients in the enterprise. There are a number of contributions of this dissertation. First, we show the practical application of conflict graphs for infrastructure-based interference management in dense wireless networks. A prototype design exhibits throughput gains of up to 50% over traditional approaches. Second, we develop novel schemes for designing a conflict graph measurement system for enterprise WLANs that can detect interference at microsecond-level timescales and with little network overhead. This allows us to compute the conflict graph up to 400 times faster as compared to the current best practice proposed in the literature. The system does not require any modifications to clients or any specialized hardware for its operation. Although the system is designed for enterprise WLANs, the proposed techniques and corresponding results are applicable to other wireless systems as well (e.g. wireless mesh networks). Third, our work opens up the space for designing novel fine-grained interference-aware protocols/algorithms that exploit the ability to compute the conflict graph at small timescales. We demonstrate an instance of such a system with the design and implementation of an architecture that dynamically manages client associations and traffic in an enterprise WLAN. We show how mobile clients sustain uninterrupted and consistent VoIP call quality in the presence of background interference for the duration of their VoIP sessions.

The traditional voice service in mobile networks is an important service that mobile users expect high quality from. With the convergence of mobile networks towards an all-IP network, an IP-based speech service becomes important which is referred to as Voice-over-IP (VoIP). The traditional voice service is highly optimized and a VoIP service must therefore fulfil strict quality requirements to provide the same speech service quality. The air interface technology, WCDMA, which is used in third generation communication systems in Europe is constantly developed. An improved concept for the mobile-to-network transmission, called the Enhanced Uplink (EUL) provides for higher uplink capacity for packet data services. It also includes features that may provide a sufficient VoIP service quality in mobile networks, when considering the uplink transmission. The purpose of this thesis is to evaluate the VoIP capacity over EUL and identify crucial aspects of radio resource management in order to increase the capacity. This is done through dynamic system simulations, using a realistic VoIP traffic model. The VoIP capacity is also estimated by a derived theoretical framework.\newline It is shown by simulation results and theoretical estimations, that power control is a vital mechanism in order to increase the capacity. Simulation results indicate that a VoIP over EUL capacity of 65\% of the traditional voice service capacity may be reached. The results also indicate that to improve the capacity for larger cells, the allowed VoIP packet delay must be increased.

Wireless networks are growing at a phenomenal rate. This growth is causing an overcrowding of the unlicensed RF spectrum, leading to increased interference between co-located devices. Existing decentralized medium access control (MAC) protocols (e.g. IEEE 802.11a/b/g standards) are poorly designed to handle interference in such dense wireless environments. This is resulting in networks with poor and unpredictable performance, especially for delay-sensitive applications such as voice and video. This dissertation presents a practical conflict-graph (CG) based approach to designing self-organizing enterprise wireless networks (or WLANs) where interference is centrally managed by the network infrastructure. The key idea is to use potential interference information (available in the CG) as an input to algorithms that optimize the parameters of the WLAN.We demonstrate this idea in three ways. First, we design a self-organizing enterprise WLAN and show how the system enhances performance over non-CG based schemes, in a high fidelity network simulator. Second, we build a practical system for conflict graph measurement that can precisely measure interference (for a given network configuration) in dense wireless environments. Finally, we demonstrate the practical benefits of the conflict graph system by using it in an optimization framework that manages associations and traffic for mobile VoIP clients in the enterprise. There are a number of contributions of this dissertation. First, we show the practical application of conflict graphs for infrastructure-based interference management in dense wireless networks. A prototype design exhibits throughput gains of up to 50% over traditional approaches. Second, we develop novel schemes for designing a conflict graph measurement system for enterprise WLANs that can detect interference at microsecond-level timescales and with little network overhead. This allows us to compute the conflict graph up to 400 times faster as compared to the current best practice proposed in the literature. The system does not require any modifications to clients or any specialized hardware for its operation. Although the system is designed for enterprise WLANs, the proposed techniques and corresponding results are applicable to other wireless systems as well (e.g. wireless mesh networks). Third, our work opens up the space for designing novel fine-grained interference-aware protocols/algorithms that exploit the ability to compute the conflict graph at small timescales. We demonstrate an instance of such a system with the design and implementation of an architecture that dynamically manages client associations and traffic in an enterprise WLAN. We show how mobile clients sustain uninterrupted and consistent VoIP call quality in the presence of background interference for the duration of their VoIP sessions.

With the development of Internet, more and more people believe that the future telecommunication network will be constructed based on IP technology. Session Initiation Protocol (SIP), which has advantages of simple entrainment method, good scalability and open protocols, is the main research topic on Voice-over-IP (VoIP). Although the client-server architecture currently used by SIP is simple and easy to maintain, it has limitation wherein service quality needs to rely on server performance. To improve this, the Internet Engineering Task Force (IETF) has created a draft to discuss the application of P2P (Peer-to-Peer) architecture in SIP, and we hope that the draft can help to provide good SIP service quality on P2P architecture, such as good fault tolerance and transmission performance. Our research is based on Chord architecture and aims to make P2P SIP architecture in an embedded User Agent. For the SIP internet telephone feature, we adjusts Chord algorithm to meet SIP internet telephone requirements. Furthermore, the adjustment to Chord makes it more applicable to the environment that users continuously join or leave, so that the revised Chord can be implemented with SIP protocol to achieve the P2P SIP goal.

The increasing demand for real-time multimedia applications and the lack of quality of service (QoS) support in public best-effort or Internet Protocol (IP) networks has prompted many researchers to propose improvements on the QoS of such networks. This research aims to improve the QoS of real-time multimedia applications in public best-effort networks, without modifying the core network infrastructure or the existing codecs of the original media applications. A source buffering control is studied based on a fluid model developed for a single flow transported over a best-effort network while allowing for flow reversal. It is shown that this control is effective for QoS improvement only when there is sufficient flow reversal or packet reordering in the network. An alternate control strategy based on predictive multi-path switching is studied where only two paths are considered as alternate options. Initially, an emulation study is performed, exploring the impact of path loss rate and traffic delay signal frequency content on the proposed control. The study reveals that this control strategy provides the best QoS improvement when the average comprehensive loss rates of the two paths involved are between 5% and 15%, and when the delay signal frequency content is around 0.5 Hz. Linear and nonlinear predictors are developed using actual network data for use in predictive multi-path switching control. The control results show that predictive path switching is better than no path switching, yet no one predictor developed is best for all cases studied. A voting based control strategy is proposed to overcome this problem. The results show that the voting based control strategy results in better performance for all cases studied. An actual voice quality test is performed, proving that predictive path switching is better than no path switching. Despite the improvements obtained, predictive path switching control has some scalability problems and other shortcomings that require further investigation. If there are more paths available to choose from, the increasing overhead in probing traffic might become unacceptable. Further, if most of the VoIP flows on the Internet use this control strategy, then the conclusions of this research might be different, requiring modifications to the proposed approach. Further studies on these problems are needed.

Voice Over IP (VoIP) systems and transmission technologies have now become the norm for many communications applications. However, whether they are used for personal communication or priority business conferences and talks, privacy and confidentiality of the communication is of utmost priority. The present industry standard is to encrypt VoIP calls using Secure Real-time Transport Protocol (SRTP), aided by ZRTP, but this methodology remains vulnerable to traffic analysis attacks, some of which utilize the length of the encrypted packets to infer the language and spoken phrases of the conversation. Secure Real-time Padding Protocol (SRPP) is a new RTP profile which pads all VoIP sessions in a unique way to thwart traffic analysis attacks on encrypted calls. It pads every RTP or SRTP packet to a predefined packet size, adds dummy packets at the end of every burst in a controllable way, adds dummy bursts to hide silence spurts, and hides information about the packet inter-arrival timings. This thesis discusses a few practical approaches and a theoretical optimization approach to packet size padding. SRPP has been implemented in the form of a library, libSRPP, for VoIP application developers and as an application, SQRKal, for regular users. SQRKal also serves as an extensive platform for implementation and verification of new packet padding techniques.

Because Ministry of Transportation and Communication want to improve its service quality and increase its revenue, Unite Communication Technology Ltd. (UCT) bring up ¡§BOT of E-Commerce and Wireless Internet¡¨ to built up the service of wireless internet and e-commerce on the train and in the station. Under the project, this thesis address to operate the value-added service of VOIP business with wireless internet to boost the total return ratio. According to data collection and survey, the financial model was erected to evaluate the feasible of this new business model. To sum up the result, the NPV of the project is 50,227,238 in 19 years (the discount rate equal to 6.49), IRR reach to 23.93%, and payback period is 7.415 years. Although the payback period is long, in terms of sustained and stable profit in the future, the project is still valuable. Furthermore, UCT is a telecommunication company so that the project can use the bandwidth more efficiently to maximum the corporation revenue. Keywords¡GWireless Internet, Voice on Internet Protocol (VOIP), Financial Feasibility Analysis

The emerging popularity and interest in Voice-over-IP (VoIP) has been accompanied by customer concerns about voice quality over these networks. The lack of an appropriate real-time capable infrastructure in packet networks along with the threats of denial-of service (DoS) attacks can deteriorate the service that these voice calls receive. And these conditions contribute to the decline in call quality in VoIP applications; therefore, error-correcting/concealing techniques remain the only alternative to provide a reasonable protection for VoIP calls against packet losses. Traditionally, each voice call employs its own end-to-end forward-error-correction (FEC) mechanisms. In this paper, we show that when VoIP calls are aggregated over a provider's link, with a suitable linear-time encoding for the aggregated voice traffic, considerable quality improvement can be achieved with little redundancy. We show that it is possible to achieve rates closer to channel capacity as more calls are combined with very small output loss rates even in the presence of significant packet loss rates in the network. The advantages of the proposed scheme far exceed similar or other coding techniques applied to individual voice calls.

With the rapid growth of communication, applications in traffic sensor network such as the multimedia service Server Push, are indispensable to business advertising. The applications of server push, for example, the service Push Mail provided by ISP (Internet Service Provider), became much more hot and popular. This paper aims to design a mobile vehicular network to let the people who received the advertisement communicate with the providers of the service. Nevertheless, the handoff problem that arises when traveling brings down the quality of communication. Therefore, seamless handoff becomes a very important issue for us to research. This thesis uses the new SCTP (Stream Control Transmission Protocol) to solve the problem. SCTP not only keeps the advantages of TCP/IP but also provides new support. Allowing the user to own several IP addresses at the same time is one of its important characteristics, which allows you to switch to any other available IP immediately if the transmission is interruptted. With this characteristic, the user barely notices any interruption in the process of handoff. Finally, we propose a new design for SCTP to achieve a response time, and can be used in applications that require a fast response time.

Workshop "Netz- und Service-Infrastrukturen" Das Inter-Asterisk eXchange (IAX)-Protokoll ermöglicht eine unproblematische Kommunikation zwischen IAX-fähigen VoIP-Systemen. In der Präsentation zu dem Vortrag werden das Protokoll vorgestellt und die Vorteile von IAX skizziert.