Network emulation focusing on QoS-Oriented satellite communication

This chapter proposes network emulation basics and a complete case study of QoS-oriented Satellite Communication

Evaluation of cross-layer reliability mechanisms for satellite digital multimedia broadcast

This paper presents a study of some reliability mechanisms which may be put at work in the context of Satellite Digital Multimedia Broadcasting (SDMB) to mobile devices such as handheld phones. These mechanisms include error correcting codes, interleaving at the physical layer, erasure codes at intermediate layers and error concealment on the video decoder. The evaluation is made on a realistic satellite channel and takes into account practical constraints such as the maximum zapping time and the user mobility at several speeds. The evaluation is done by simulating different scenarii with complete protocol stacks. The simulations indicate that, under the assumptions taken here, the scenario using highly compressed video protected by erasure codes at intermediate layers seems to be the best solution on this kind of channel

Overcoming TCP Degradation in the Presence of Multiple Intermittent Link Failures Utilizing Intermediate Buffering

It is well documented that assumptions made in the popular Transmission Control Protocol\u27s (TCP) development, while essential in the highly reliable wired environment, are incompatible with today\u27s wireless network realities in what we refer to as a challenged environment. Challenged environments severely degrade the capability of TCP to establish and maintain a communication connection with reasonable throughput. This thesis proposes and implements an intermediate buffering scheme, implemented at the transport layer, which serves as a TCP helper protocol for use in network routing equipment to overcome short and bursty, but regular, link failures. Moreover, the implementation requires no modifications to existing TCP implementations at communicating nodes and integrates well with existing routing equipment. In a simulated six-hop network with five modified routers supporting four challenged links, each with only 60% availability, TCP connections are reliably established and maintained, despite the poor link availability, whereas 94% fail using standard routing equipment, i.e., without the TCP helper protocol

Performance Study of Shared Versus Nonshared Bandwidth on a Packet-Switched Network

In wide area computer data communications, many networks have evolved by satisfying increased user demands in the most expedient manner. In some cases, new users’ demands are satisfied by installing a new link, rather than sharing the links that are already in place. This research investigates the differences in performance between using a dedicated link for each source destination pair (nonshared bandwidth) and using a single link to be used by all source destination pairs (shared bandwidth). Simulation models are developed for a wide area network using shared bandwidth, and a wide area network using nonshared bandwidth. The quality of service offered by each network is based on its responsiveness and productivity. Responsiveness will be measured in terms of average end to end delay of packet transmission, and productivity will be measured in terms of percent bandwidth utilization. The networks are modeled under a common set of operating assumptions and system environment. This allows for accurate comparison of packet delay and bandwidth utilization. Two variable input parameters are used in the simulation: intensity of input traffic load, and amount of link capacity. Provided that the intensity of the input traffic load remains below the network saturation level, it is shown that the shared system clearly outperforms the nonshared system. This result occurs for both a uniform and a nonuniform traffic load destination distribution

Mobile-IP ad-hoc network MPLS-based with QoS support.

The support for Quality of Service (QoS) is the main focus of this thesis. Major issues and challenges for Mobile-IP Ad-Hoc Networks (MANETs) to support QoS in a multi-layer manner are considered discussed and investigated through simulation setups. Different parameters contributing to the subjective measures of QoS have been considered and consequently, appropriate testbeds were formed to measure these parameters and compare them to other schemes to check for superiority. These parameters are: Maximum Round-Trip Delay (MRTD), Minimum Bandwidth Guaranteed (MBG), Bit Error Rate (BER), Packet Loss Ratio (PER), End-To-End Delay (ETED), and Packet Drop Ratio (PDR) to name a few. For network simulations, NS-II (Network Simulator Version II) and OPNET simulation software systems were used.Dept. of Electrical and Computer Engineering. Paper copy at Leddy Library: Theses & Major Papers - Basement, West Bldg. / Call Number: Thesis2005 .A355. Source: Masters Abstracts International, Volume: 44-03, page: 1444. Thesis (M.Sc.)--University of Windsor (Canada), 2005

Streaming Video Performance and Enhancements in Resource-Constrained Wireless Networks

Streaming video is an increasingly popular application in wireless networks. The concept of a live streaming video yields several enticing possibilities: real-time video conferencing, television broadcasting, pay-per-view movie streaming, and more. These ideas have already been explored via the internet and have met with mixed success, largely due to the shortcomings of the underlying network. Taking streaming video to wireless networks, then, poses several significant challenges. Wireless networks are inherently more susceptible to failures and data corruption due to their unstable communications medium. This volatility suggests serious drawbacks for any implementation of streaming video. Video frame errors, jitter, and even complete sync loss are entirely conceivable in a wireless environment. Many of these issues have been undertaken and several approaches to mediation or even solution of these problems are underway. This thesis proposes to use advanced simulation techniques to properly exhaustively permute many vital parameters within a UMTS network and uncover, if they exist, bottlenecks in UMTS performance under considerable network load. This is accomplished via a described testing plan with simulation environment. Additionally this thesis proposes a new UDP-like transport layer specially optimized for streaming media over resource-constrained networks, tested to work with significant improvements under the UMTS cellular networking system. Finally this thesis provides several innovative new methods in the furtherance of the field of streaming media research in resourceconstrained and cellular environments. Overall this thesis makes several important contributes to an exciting and ever-growing field of active research and discussion

Supporting Real-Time Applications in an Integrated Services Packet Network: Architecture and Mechanism

This paper considers the support of real-time applications in an Integrated Services Packet Network (ISPN). We first review the characteristics of real-time applications. We observe that, contrary to the popular view that real-time applications necessarily require a fixed delay bound, some real-time applications are more flexible and can adapt to current network conditions. We then propose an ISPN architecture that supports two distinct kinds of real-time service: guaranteed service, which is the traditional form of real-time service discussed in most of the literature and involves pre-computed worst-case delay bounds, and predicted service, which uses the measured performance of the network in computing delay bounds. We then propose a packet scheduling mechanism that can support both of these real-time services as well as accommodate datagram traffic. We also discuss two other aspects of an overall ISPN architecture: the service interface and the admission control criteria.Research at MIT was supported by DARPA through NASA Grant NAG 2-582, by NSF grant NCR-8814187, and by DARPA and NSF through Cooperative Agreement NCR-8919038 with the Corporation for National Research Initiatives