Issues in providing a reliable multicast facility

Issues involved in point-to-multipoint communication are presented and the literature for proposed solutions and approaches surveyed. Particular attention is focused on the ideas and implementations that align with the requirements of the environment of interest. The attributes of multicast receiver groups that might lead to useful classifications, what the functionality of a management scheme should be, and how the group management module can be implemented are examined. The services that multicasting facilities can offer are presented, followed by mechanisms within the communications protocol that implements these services. The metrics of interest when evaluating a reliable multicast facility are identified and applied to four transport layer protocols that incorporate reliable multicast

A Novel Network Coded Parallel Transmission Framework for High-Speed Ethernet

Parallel transmission, as defined in high-speed Ethernet standards, enables to use less expensive optoelectronics and offers backwards compatibility with legacy Optical Transport Network (OTN) infrastructure. However, optimal parallel transmission does not scale to large networks, as it requires computationally expensive multipath routing algorithms to minimize differential delay, and thus the required buffer size, optimize traffic splitting ratio, and ensure frame synchronization. In this paper, we propose a novel framework for high-speed Ethernet, which we refer to as network coded parallel transmission, capable of effective buffer management and frame synchronization without the need for complex multipath algorithms in the OTN layer. We show that using network coding can reduce the delay caused by packet reordering at the receiver, thus requiring a smaller overall buffer size, while improving the network throughput. We design the framework in full compliance with high-speed Ethernet standards specified in IEEE802.3ba and present solutions for network encoding, data structure of coded parallel transmission, buffer management and decoding at the receiver side. The proposed network coded parallel transmission framework is simple to implement and represents a potential major breakthrough in the system design of future high-speed Ethernet.Comment: 6 pages, 8 figures, Submitted to Globecom201

Development of a Reliable Multicast Protocol in Mobile Ad Hoc Networks

Mobile ad hoc network is a collection of mobile nodes forming dynamic and temporary network. The mobile nodes work in collaborative nature to carry out a given task. It can receive and transmit data packets without the use of any existing network infrastructure or centralized administration. Multicasting is among the pertinent issues of communication in such networks. The reliable delivery of multicast data packets needs feedback from all multicast receivers to indicate whether a retransmission is necessary. The Feedback Implosion Problem (FIP) states that reliable multicast in ad hoc networks suffers from redundant feedback packets, loss, duplication, and out-of-order delivery of data packets. To carry out this task, several reliable multicast protocols have been proposed to reduce the number of feedback packets from the receiver nodes. This is achieved by placing the responsibility to detect packet loss and initiating loss recovery timer on the receiver nodes which is complemented by feedback suppression. The initiating loss recovery timer depends on the number of hops between the nodes. As the dynamic nature of the number of hops between the nodes in ad hoc networks is unstable the loss recovery timer become inaccurate. Thus, the inaccuracy of the loss recovery timer, in return, causes extra overhead and more delays. The main objectives of this research are to enhance the FIP and decrease the recovery delays in reliable multicast protocol for mobile ad hoc networks using suggested approaches. First, the Source Tree Reliable Multicast (STRM) protocol adopting a novel technique to select a subset of one-hop neighbors from the sender node as its Forward Servers (FS). The key idea behind selecting this subset one-hop neighbors is to forward the retransmitted lost data packets and to receive the feedback packets from the receiver nodes. Second, proposed two algorithms to improve the performance of the STRM protocol. The first algorithm is developed to avoid the buffer overflow in the FS nodes. This is achieved by managing the buffer of the FS nodes; by selecting the FS nodes depending on the empty buffer size it has and reducing the amount of feedback sent from the receiver nodes to their FS node. The second algorithm is developed to decrease the number of duplicated packets in the multicast members in the local group. This is achieved by sending the repair packets only to the member that has requested it. The FS in the local group should create a dynamic and temporary sub group whose members are only the members that requested the retransmission of the repair packet. The approaches were tested using detailed discrete-event simulation model which was developed encompassing messaging system that includes error, delay and mobility models to characterize the performance benefits of the proposed algorithms in comparison to ReMHoc protocol. Our approaches achieve up to 2.19% improvement on average packet delivery ratio, 3.3% on requested packets, and 46% on recovery latency time without incurring any additional communication or intense computation

Analysis domain model for shared virtual environments

The field of shared virtual environments, which also encompasses online games and social 3D environments, has a system landscape consisting of multiple solutions that share great functional overlap. However, there is little system interoperability between the different solutions. A shared virtual environment has an associated problem domain that is highly complex raising difficult challenges to the development process, starting with the architectural design of the underlying system. This paper has two main contributions. The first contribution is a broad domain analysis of shared virtual environments, which enables developers to have a better understanding of the whole rather than the part(s). The second contribution is a reference domain model for discussing and describing solutions - the Analysis Domain Model

Issues in designing transport layer multicast facilities

Multicasting denotes a facility in a communications system for providing efficient delivery from a message's source to some well-defined set of locations using a single logical address. While modem network hardware supports multidestination delivery, first generation Transport Layer protocols (e.g., the DoD Transmission Control Protocol (TCP) (15) and ISO TP-4 (41)) did not anticipate the changes over the past decade in underlying network hardware, transmission speeds, and communication patterns that have enabled and driven the interest in reliable multicast. Much recent research has focused on integrating the underlying hardware multicast capability with the reliable services of Transport Layer protocols. Here, we explore the communication issues surrounding the design of such a reliable multicast mechanism. Approaches and solutions from the literature are discussed, and four experimental Transport Layer protocols that incorporate reliable multicast are examined

WAIT: Selective Loss Recovery for Multimedia Multicast.

Recently the Internet has been increasingly used for multi-party applications like video-conferencing, video-on-demand and shared white-boards. Multicast extensions to IP to support multi-party applications are best effort, often resulting in packet loss within the network. Since some multicast applications can not tolerate packet loss, most of the existing reliable multicast schemes recover each and every lost packet. However, multimedia applications can tolerate a certain amount of packet loss and are sensitive to long recovery delays. We propose a new loss recovery technique that selectively repairs lost packets based upon the amount of packet loss and delay expected for the repair. Our technique sends a special WAIT message down the multicast tree in the event a loss is detected in order to reduce the number of retransmission requests. We also propose an efficient sender initiated multicast trace-route mechanism for determining the multicast topology and a mechanism to deliver the topology information to the multicast session participants. We evaluate our proposed technique using an event driven network simulator, comparing it with two popular reliable multicast protocols, SRM and PGM. We conclude that our proposed WAIT protocol can reduce the overhead on a multicast session as well as improve the average end-to-end latency of the session

Improving the Scalability of DPWS-Based Networked Infrastructures

The Devices Profile for Web Services (DPWS) specification enables seamless discovery, configuration, and interoperability of networked devices in various settings, ranging from home automation and multimedia to manufacturing equipment and data centers. Unfortunately, the sheer simplicity of event notification mechanisms that makes it fit for resource-constrained devices, makes it hard to scale to large infrastructures with more stringent dependability requirements, ironically, where self-configuration would be most useful. In this report, we address this challenge with a proposal to integrate gossip-based dissemination in DPWS, thus maintaining compatibility with original assumptions of the specification, and avoiding a centralized configuration server or custom black-box middleware components. In detail, we show how our approach provides an evolutionary and non-intrusive solution to the scalability limitations of DPWS and experimentally evaluate it with an implementation based on the the Web Services for Devices (WS4D) Java Multi Edition DPWS Stack (JMEDS).Comment: 28 pages, Technical Repor

System Support for Bandwidth Management and Content Adaptation in Internet Applications

This paper describes the implementation and evaluation of an operating system module, the Congestion Manager (CM), which provides integrated network flow management and exports a convenient programming interface that allows applications to be notified of, and adapt to, changing network conditions. We describe the API by which applications interface with the CM, and the architectural considerations that factored into the design. To evaluate the architecture and API, we describe our implementations of TCP; a streaming layered audio/video application; and an interactive audio application using the CM, and show that they achieve adaptive behavior without incurring much end-system overhead. All flows including TCP benefit from the sharing of congestion information, and applications are able to incorporate new functionality such as congestion control and adaptive behavior.Comment: 14 pages, appeared in OSDI 200