47 research outputs found

    Performance evaluation of MPEG-4 video streaming over UMTS networks using an integrated tool environment

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    Universal Mobile Telecommunications System (UMTS) is a third-generation mobile communications system that supports wireless wideband multimedia applications. This paper investigates the video quality attained in streaming MPEG-4 video over UMTS networks using an integrated tool environment, which comprises an MPEG-4 encoder/decoder, a network simulator and video quality evaluation tools. The benefit of such an integrated tool environment is that it allows the evaluation of real video sources compressed using an MPEG-4 encoder. Simulation results show that UMTS Radio Link Control (RLC) outperforms the unacknowledged mode. The latter mode provides timely delivery but no error recovery. The acknowledged mode can deliver excellent perceived video quality for RLC block error rates up to 30% utilizing a playback buffer at the streaming client. Based on the analysis of the performance results, a self-adaptive RLC acknowledged mode protocol is proposed

    Convergence of packet communications over the evolved mobile networks; signal processing and protocol performance

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    In this thesis, the convergence of packet communications over the evolved mobile networks is studied. The Long Term Evolution (LTE) process is dominating the Third Generation Partnership Project (3GPP) in order to bring technologies to the markets in the spirit of continuous innovation. The global markets of mobile information services are growing towards the Mobile Information Society. The thesis begins with the principles and theories of the multiple-access transmission schemes, transmitter receiver techniques and signal processing algorithms. Next, packet communications and Internet protocols are referred from the IETF standards with the characteristics of mobile communications in the focus. The mobile network architecture and protocols bind together the evolved packet system of Internet communications to the radio access network technologies. Specifics of the traffic models are shortly visited for their statistical meaning in the radio performance analysis. Radio resource management algorithms and protocols, also procedures, are covered addressing their relevance for the system performance. Throughout these Chapters, the commonalities and differentiators of the WCDMA, WCDMA/HSPA and LTE are covered. The main outcome of the thesis is the performance analysis of the LTE technology beginning from the early discoveries to the analysis of various system features and finally converging to an extensive system analysis campaign. The system performance is analysed with the characteristics of voice over the Internet and best effort traffic of the Internet. These traffic classes represent the majority of the mobile traffic in the converged packet networks, and yet they are simple enough for a fair and generic analysis of technologies. The thesis consists of publications and inventions created by the author that proposed several improvements to the 3G technologies towards the LTE. In the system analysis, the LTE showed by the factor of at least 2.5 to 3 times higher system measures compared to the WCDMA/HSPA reference. The WCDMA/HSPA networks are currently available with over 400 million subscribers and showing increasing growth, in the meanwhile the first LTE roll-outs are scheduled to begin in 2010. Sophisticated 3G LTE mobile devices are expected to appear fluently for all consumer segments in the following years

    Satellite-3G Hybrid Networks: Impact of ACM and ARQ on TCP Performance

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    The adoption of satellite systems in providing broadband transmissions to mobile users such as trains, buses and vans is expected to be an interesting solution. The scenariowe considered refers to a hybrid network architecture, where a geostationary satellite forward link and a terrestrial 3G return link are used in order to exploit both the high bandwidth of a satellite channel and the lower propagation delay of a terrestrial path. The resulting round-trip delay is much shorter than that one experienced by using both the forward and return link via satellite. This is particularly appealing for overcoming the TCP efficiency degradation in high delay-bandwidth product and error prone channels. In this hybrid scenario, we used simulation results to compare the goodput of four of the most popular TCP variants, in the presence of a GOOD-BAD satellite channel, as the one experienced by mobile users. We applied an AdaptiveCoding and Modulation (ACM) technique as well, and studied its impact on TCP efficiency, when used both alone and in cooperation with an Automatic Repeat reQuest (ARQ) scheme of the Selective Repeat (SR) type with low persistency. Results obtained indicate that this hybrid architecture is advantageous for TCP transmissions in terms of average goodput, and that ACM is effective only if it is jointly used with ARQ schemes

    Quality of service and resource management in IP and wireless networks

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    A common theme in the publications included in this thesis is the quality of service and resource management in IP and wireless networks. This thesis presents novel algorithms and implementations for admission control in IP and IEEE 802.16e networks, active queue management in EGPRS, WCDMA, and IEEE 802.16e networks, and scheduling in IEEE 802.16e networks. The performance of different algorithms and mechanisms is compared with the prior art through extensive ns-2 simulations. We show that similar active queue management mechanisms, such as TTLRED, can be successfully used to reduce the downlink delay (and in some cases even improve the TCP goodput) in different bottlenecks of IP, EGPRS, WCDMA, and IEEE 802.16e access networks. Moreover, almost identical connection admission control algorithms can be applied both in IP access networks and at IEEE 802.16e base stations. In the former case, one just has to first gather the link load information from the IP routers. We also note that DiffServ can be used to avoid costly overprovisioning of the backhaul in IEEE 802.16e networks. We present a simple mapping between IEEE 802.16e data delivery services and DiffServ traffic classes, and we propose that IEEE 802.16e base stations should take the backhaul traffic load into account in their admission control decisions. Moreover, different IEEE 802.16e base station scheduling algorithms and uplink channel access mechanisms are studied. In the former study, we show that proportional fair scheduling offers superior spectral efficiency when compared to deficit round-robin, though in some cases at the cost of increased delay. Additionally, we introduce a variant of deficit round-robin (WDRR), where the quantum value depends on the modulation and coding scheme. We also show that there are several ways to implement ertPS in an efficient manner, so that during the silence periods of a VoIP call no uplink slots are granted. The problem here, however, is how to implement the resumption after the silence period while introducing as little delay as possible

    Transport Layer Optimizations for Heterogeneous Wireless Multimedia Networks

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    The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.Ph.D.Committee Chair: Vijay Madisetti; Committee Member: Raghupathy Sivakumar; Committee Member: Sudhakar Yalamanchili; Committee Member: Umakishore Ramachandran; Committee Member: Yucel Altunbasa

    A study of mobile VoIP performance in wireless broadband networks

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    Voice service is to date still the killer mobile service and the main source for operator revenue for years to come. Additionally, voice service will evolve from circuit switched technologies towards packet based Voice over IP (VoIP). However, using VoIP over wireless networks different from 3GPP cellular technologies makes it also a disruptive technology in the traditional telecommunication sector. The focus of this dissertation is on determining mobile VoIP performance in different wireless broadband systems with current state of the art networks, as well as the potential disruption to cellular operators when mobile VoIP is deployed over different access networks. The research method is based on an empirical model. The model and experiments are well documented and based on industry standards for voice quality evaluation. The evaluation provides results from both experiments in a controlled laboratory setup as well as from live scenarios. The research scope is first, evaluate each network technology independently; second, investigate vertical handover mobility cases; third, determine other aspects directly affecting end user experience (e.g., call setup delay and battery lifetime). The main contribution of this work is a systematic examination of mobile VoIP performance and end user experience. The research results point out the main challenges for achieving call toll quality, and how derive the required changes and technological performance roadmap for improved VoIP service. That is, investigate how the performance and usability of mobile VoIP can eventually be improved to be a suitable substitute for circuit switched voice. In addition, we evaluate the potential disruption to cellular operators that mobile VoIP brings when deployed over other access networks. This research extends the available knowledge from simulations and provides an insight into actual end user experience, as well as the challenges of using embedded clients in handheld devices. In addition, we find several issues that are not visible or accounted for in simulations in regard to network parameters, required retransmissions and decreased battery lifetime. The conclusion is that although the network performance of several wireless networks is good enough for near toll quality voice in static scenarios, there are still a number of problems which make it currently unfeasible to use as a primary voice service. Moreover, under mobility scenarios performance is degraded. Finally, there are other issues apart from network performance such as energy consumption, hardware limitations and lack of supporting business models (e.g., for WiFi mesh) that further limit the possibility of rolling out mobile VoIP services
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