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No / Providing a guaranteed quality-of-service (QoS) is essential to many real-time applications. The existing distributed QoS routing algorithms are based on either shortest path or flooding and both tend to have high message overhead. A new distributed unicast QoS routing algorithm based on Fano¿s decoding method is studied. Fano¿s decoding method is a technique used in error control coding and it attempts to trace an optimal path probabilistically. The similarity of various aspects of Fano¿s decoding method to a QoS routing algorithm and the benefits it can provide encourages us to investigate the possibility of using it in QoS routing. This is the first known attempt to modify an error control technique using Fano¿s decoding method for the purpose of QoS routing in fixed wired networks. Simulation results demonstrate the effectiveness of the proposed algorithm in terms of message overhead and success ratio (% of paths obtained that satisfy the given QoS constraints). It is shown that the message overhead in the proposed algorithm is reduced compared to flooding based algorithms while maintaining a similar success ratio. Message overhead is also reduced compared to distance vector based algorithms for all but very sparsely connected networks, while success ratio is not compromised. Nodal storage is also considerably lower than for most other contemporary QoS routing algorithms.

No / Quality of Service (QoS) is of extreme importance in accommodating the increasingly diverse range of services and types of traffic in present day communication networks and delay is one of the most important QoS metrics. This paper presents a new approach for constraining queueing delay in a buffer to a specified level as the arrival rate changes with time. A discrete-time control algorithm is presented that operates on a buffer (queue) which incorporates a moveable threshold. An algorithm is developed that controls the delay by dynamically adjusting the threshold which, in turn, controls the arrival rate. The feasibility of the system is examined using both theoretical analysis and simulation.

The lack of control over the distribution network when using IP-based delivery methods for audio content, introduces challenges in maintaining Quality of Experience (QoE). This thesis investigates the impact of network conditions on the QoE for live radio broadcasts from Sveriges Radio. Through the utilization of a prototype that analyses a HLS (HTTP Live Streaming) stream, data was collected and analyzed to understand the impact of different metrics for broadcast stability and QoE. Different network conditions were measured and simulated via a network throttler. Findings reveal that a minimum channel capacity threshold is necessary to maintain a stable broadcast without rebuffering events. Important Key Performance Indicators (KPIs) for QoE, including bandwidth, throughput, rebuffering events, and audio bitrate qualitychanges, wereidentified. Additionally, atraceroutepathanalysisidentified a specific router as the bottleneck within the delivery chain. / Bristen på kontroll över distributionsnätverket vid användning av IP-baserade leveransmetoder för ljud, innebär utmaningar för att upprätthålla Quality of Experience (QoE). Denna rapport undersöker effekten av nätverks förhållanden på QoE för direktsända radioutsändningar från Sveriges Radio. Genom en prototyp som analyserar en HLS (HTTP Live Streaming) ström samlades data och analyserades för att förstå effekterna av olika mättvärden försändningsstabilitet och QoE. Olika nätverks förhållanden mättes och simulerades via en nätverksdämpare. Resultaten visar att en minimal tröskel i kanalkapacitet är nödvändig för att upprätthålla en stabil sändning utan buffer. Viktiga Key Performance Indicators (KPIs) för QoE identifierades, blanda annat bandbredd, genomströmning, buffring och ändringar i ljudets kvalitet. Dessutom gjordes en traceroute-analys där en specifik router identifierades som en flaskhal inom leveranskedjan.

In the recent decade, peer-to-peer (P2P) technology has greatly enhanced the scalability of multimedia streaming on the Internet by enabling efficient cooperation among end-users. However, existing streaming applications are plagued by the problems of long playback latency and long churn-induced delays. First of all, many streaming applications, such as IPTV and video conferencing, have rigorous constraints on end-to-end delays. Moreover, churn-induced delays, including delays from channel switching and streaming recovery, in current P2P streaming applications are typically in the scale of 10-60 seconds, which is far below the favorable user experience as in cable TV systems. These two issues in terms of playback latency and churn-induced delays have hindered the extensive commercial deployment of P2P systems. Motivated by this, in this dissertation, we focus on reducing delays in P2P live streaming systems. Specifically, we propose solutions for reducing delays in P2P live streaming systems in four problem spaces: (1) minimizing the maximum end-to-end delay in P2P streaming; (2) minimizing the average end-to-end delay in P2P streaming; (3) minimizing the average delay in multi-channel P2P streaming; and (4) reducing churn-induced delays. We devise a streaming scheme to minimize the maximum end-to-end streaming delay under a mesh-based overlay network paradigm. We call this problem, the MDPS problem. We formulate the MDPS problem and prove its NP-completeness. We then present a polynomial-time approximation algorithm, called Fastream-I, for this problem, and show that the performance of Fastream-I is bounded by a ratio of O(SQRT(log n)), where n is the number of peers in the system. We also develop a distributed version of Fastream-I that can adapt to network dynamics. Our simulation study reveals the effectiveness of Fastream-I, and shows a reasonable message overhead. While Fastream-I yields the minimum maximum end-to-end streaming delay (within a factor of O(SQRT(log n)), in many P2P settings, users may desire the minimum average end-to-end P2P streaming delay. Towards this, we devise a streaming scheme which optimizes the bandwidth allocation to achieve the minimum average end-to-end P2P streaming delay. We call this problem, the MADPS problem. We first develop a generic analytical framework for the MADPS problem. We then present Fastream-II as a solution to the MADPS problem. The core part of Fastream-II is a fast approximation algorithm, called APX-Fastream-II, based on primal-dual schema. We prove that the performance of APX-Fastream-II is bounded by a ratio of 1+w, where w is an adjustable input parameter. Furthermore, we show that the flexibility of w provides a trade-off between the approximation factor and the running time of Fastream-II. The third problem space of the dissertation is minimizing the average delay in multi-channel P2P streaming systems. Toward this, we present an algorithm, called Fastream-III. To reduce the influence from frequent channel-switching behavior, we build Fastream-III for the view-upload decoupling (VUD) model, where the uploaded content from a serving node is independent of the channel it views. We devise an approximation algorithm based on primal-dual schema for the critical component of Fastream-III, called APX-Fastream-III. In contrast to APX-Fastream-II, APX-Fastream-III addresses the extra complexity in the multichannel scenario and maintains the approximation bound by a ratio of 1+w. Besides playback lag, delays occurring in P2P streaming may arise from two other factors: node churn and channel switching. Since both stem from the re-connecting request in churn, we call them churn-induced delays. Optimizing churn-induced delays is the dissertation's fourth problem space. Toward this, we propose NAP, a novel agent-based P2P scheme, that provides preventive connections to all channels. Each channel in NAP selects powerful peers as agents to represent the peers in the channel to minimize control and message overheads. Agents distill the bootstrapping peers with superior bandwidth and lifetime expectation to quickly serve the viewer in the initial period of streaming. We build a queueing theory model to analyze NAP. Based on this model, we numerically compare NAP's performance with past efforts. The results of the numerical analysis reveal the effectiveness of NAP. / Ph. D.

This research investigated the efficient utilization of wireless bandwidth in Code Division Multiple Access (CDMA)systems that support multiple data rates with Orthogonal Variable Spreading Factor (OVSF)codes. The specific problem being addressed was that currently proposed public-domain algorithms for assigning OVSF codes make inefficient use of wireless bandwidth for bursty data traffic sources with different Quality of Service (QoS) requirements. The purpose of this research was to develop an algorithm for the assignment of OVSF spreading codes in a Third-Generation (3G)Wideband CDMA (WCDMA)system. The goal of this algorithm was to efficiently utilize limited, wireless resources for bursty data traffic sources with different QoS requirements. The key contribution of this research was the implementation and testing of two code sharing techniques which are not implemented in existing OVSF code assignment algorithms. These techniques were termed statistical multiplexing and dynamic code sharing. The statistical multiplexing technique used a shared channel to support multiple bursty traffic sources. The dynamic code sharing technique supported multiple data users by temporarily granting access to dedicated channels. These techniques differed in terms of both complexity and performance guarantees. / Ph. D.

Modern mobile communication systems will provide enhanced high-speed data, multimedia, and voice services to mobile users. The integration of such heterogeneous traffic types implies that the network must provide differentiated Quality of Service (QoS). Beam forming techniques have been proposed to increase the spectral efficiency of the wireless channel. Using beamforming in the network will lead to intra-cell handoffs within the cell due to user mobility. In a commercially deployed next generation cellular system, it is likely that beam forming and QoS guarantees to the users will co-exist. In this work we propose a resource allocation and management scheme tailored for a network that employs smart antennas in support of a heterogeneous user mix. Resource management in a wireless system should take care of channel impairments and non-ideal antenna patterns. Mobile users moving from one beam to the other give rise to resource reallocation issues. Depending on the scatterer distribution in the cell, the Angle of Arrival (AoA) of the users will also change, affecting the interference pattern in the cell. In a system with data and multimedia users, some of the users are likely to be elastic in their demands for bandwidth. In this work, we propose a resource allocation and management scheme tailored for systems with smart antennas having heterogeneous users. The algorithm works by comparing the received power in the beams. Elasticity of user requirement for data services is exploited to provide adaptive QoS, thereby reducing the call dropping probability due to user mobility. Simulation results showing the channel and Multiple Access Interference (MAI) effects on system performance are presented. The effect of channel coding to provide Bit Error Rate (BER) guarantees is studied. We also show the throughput advantage obtained using the resource management algorithms. It is also seen that the throughput of the system increases for a user population having a higher elasticity. A modified resource allocation algorithm to reduce the blocking probability of the calls is presented and performance verified using simulation. The probability of call dropping in an unmanaged system due to user mobility is shown. Our studies show that using managed system the call drop probability can be minimized. / Master of Science

This thesis proposes a multi-protocol label switching (MPLS)-based architecture to provide quality of service (QoS) for both internet service provider (ISP) networks and backbone Internet Protocol (IP) networks that are heterogeneous in nature. Heterogeneous networks are present due to the use of different link-layer mechanisms in the current Internet. Copper-based links, fiber-based links, and wireless links are some examples of different physical media that lead to different link-layer mechanisms. The proposed architecture uses generalized MPLS and other MPLS features to combat heterogeneity. The proposed architecture leverages the QoS capabilities of asynchronous transfer mode (ATM) and the scalability advantages of the IP differentiated services (DiffServ) architecture. This architecture is constructed in such a way that MPLS interacts with DiffServ in the backbone networks while performing ATM-like QoS enforcement in the periphery of the networks. The architecture supports traffic engineering through MPLS explicit paths. MPLS network management, bandwidth broker capabilities, and customizability is handled through domain specific MPLS management entities that use the Common Open Policy Service (COPS) protocol to interact with other MPLS entities like MPLS label switch routers and label edge routers. The thesis provides a description of MPLS and QoS, followed by a discussion of the motivation for a new architecture. The MPLS-based architecture is then discussed and compared against similar architectures. To integrate the ATM and DiffServ QoS attributes into this architecture, MPLS signaling protocols are used. There are two common MPLS signaling protocols. They are Resource Reservation Protocol with traffic engineering extensions (RSVP-TE) and Constraint-Routed Label Distribution Protocol (CR-LDP). Both these protocols offer comparative MPLS features for constraint routed label switch path construction, maintenance, and termination. RSVP-TE uses UDP and IP, while CR-LDP uses TCP. This architecture proposes a multi-level domain of operation where CR-LDP operates in internet service provider (ISP) networks and RSVP- TE operates in backbone networks along with DiffServ. Qualitative analysis for this choice of domain of operation of the signaling protocols is then presented. Quantitative analysis through simulation demonstrates the advantages of combining DiffServ and MPLS in the backbone. The simulation setup compares the network performance in handling mixed ill-behaved and well-behaved traffic in the same link, with different levels of DiffServ and MPLS integration in the network. The simulation results demonstrate the advantages of integrating the QoS features of DiffServ, ATM functionality, and MPLS into a single architecture. / Master of Science