In distributed processing for communication services, a proper server selection scheme is required to reduce delay by ensuring the event occurrence order. Although a conservative synchronization algorithm (CSA) has been used to achieve this goal, an optimistic synchronization algorithm (OSA) can be feasible for synchronizing distributed systems. In comparison with CSA, which reproduces events in occurrence order before processing applications, OSA can be feasible to realize low delay communication as the processing events arrive sequentially. This paper proposes an optimal server selection scheme that uses OSA for distributed processing systems to minimize end-to-end delay under the condition that maximum status holding time is limited. In other words, the end-to-end delay is minimized based on the allowed rollback time, which is given according to the application designing aspects and availability of computing resources. Numerical results indicate that the proposed scheme reduces the delay compared to the conventional scheme.

This paper proposes an efficient server selection scheme in successive participation scenario with participating-domain segmentation. The scheme is utilized by distributed processing systems for real-time interactive communication to suppress the communication latency of a wide-area network. In the proposed scheme, users participate for server selection one after another. The proposed scheme determines a recommended server, and a new user selects the recommended server first. Before each user participates, the recommended servers are determined assuming that users exist in the considered regions. A recommended server is determined for each divided region to minimize the latency. The new user selects the recommended available server, where the user is located. We formulate an integer linear programming problem to determine the recommended servers. Numerical results indicate that, at the cost additional computation, the proposed scheme offers smaller latency than the conventional scheme. We investigate different policies to divide the users' participation for the recommended server finding process in the proposed scheme.

As a flexible and cost-efficient scalable Internet access network, we studied architectures, protocols, and design optimizations of the Wireless Internet-access Mesh NETwork (WIMNET). WIMNET is composed of multiple access points (APs) connected through multihop wireless communications on IEEE 802.11 standards. The increasing popularity of real-time applications such as IP-phones and IP-TV means that they should be supported in WIMNET. However, the contention resolution mechanism using a random backoff-time in the CSMA/CA protocol of 802.11 standards is not sufficient for handling real-time traffic in multihop wireless communications. In this paper, we propose a Fixed Backoff-time Switching (FBS) method for the CSMA/CA protocol to improve the real-time traffic performance in WIMNET by giving the necessary activation chances to each link. We implement our proposal on the QualNet simulator, and verify its effectiveness through simulations on three network topologies with four scenarios.

This paper proposes multipoint-to-multipoint (MPtoMP) real-time broadcast transmission using network coding for ad-hoc networks like video game networks. We aim to achieve highly reliable MPtoMP broadcasting using IEEE 802.11 media access control (MAC) that does not include a retransmission mechanism. When each node detects packets from the other nodes in a sequence, the correctly detected packets are network-encoded, and the encoded packet is broadcasted in the next sequence as a piggy-back for its native packet. To prevent increase of overhead in each packet due to piggy-back packet transmission, network coding vector for each node is exchanged between all nodes in the negotiation phase. Each user keeps using the same coding vector generated in the negotiation phase, and only coding information that represents which user signal is included in the network coding process is transmitted along with the piggy-back packet. Our simulation results show that the proposed method can provide higher reliability than other schemes using multi point relay (MPR) or redundant transmissions such as forward error correction (FEC). We also implement the proposed method in a wireless testbed, and show that the proposed method achieves high reliability in a real-world environment with a practical degree of complexity when installed on current wireless devices.

The performance of a real-time networked application can be drastically affected by delays in packets traversing the network. Some real-time applications impose limits for acceptable network delay, and so a packet which is delayed longer than the limit before arriving at its destination is worthless to the flow to which the packet belongs. Not only that, but the rejected packet is also damaging to the quality of other flows in the network, because it may increase the queuing delay for other packets. Therefore, this paper proposes an adaptive scheme using two mechanisms, in which packets experiencing too great a delay are discarded at intermediate nodes based on the delay limit for the application and the delay experienced by each packet. This earlier discarding of packets is expected to improve the overall delay performance of real-time flows competing for network resources when the network is congested. An extensive simulation is conducted, and the results show that the scheme has great potential in improving the delay performance of real-time traffic in both homogeneous and heterogeneous environments in terms of traffic volume and application delay requirements.

Packet loss and delay cause degradation in the quality of real-time, interactive applications such as video conferencing. Forward error correction (FEC) schemes have been proposed to make the applications more resilient to packet loss, because the time required to recover the lost packets is shorter than that required to retransmit the lost packets. On the other hand, the codec generally used in real-time applications like MPEG4 has the feature that the sending bit rate and the packet size of the traffic vary significantly according to the motion of an object in a video. If the traditional FEC coding, which is calculated on the basis of a fixed-size block, is applied to such applications, a waste of bandwidth and a delay variation are caused and the quality is degraded. In this paper, we propose suitable FEC schemes for visual communication systems using variable bit-rate (VBR) codec and evaluate the effectiveness of these schemes using our prototype implementation and experimental network.

Real-time applications are indispensable for conducting research and business in government, industry, and academic organizations. Recently, real-time applications with security requirements increasingly emerged in large-scale distributed systems such as Grids. However, the complexities and specialties of diverse security mechanisms dissuade users from employing existing security services for their applications. To effectively tackle this problem, in this paper we propose a security middleware (SMW) model from which security-sensitive real-time applications are enabled to exploit a variety of security services to enhance the trustworthy executions of the applications. A quality of security control manager (QSCM), a centerpiece of the SMW model, has been designed and implemented to achieve a flexible trade-off between overheads caused by security services and system performance, especially under situations where available resources are dynamically changing and insufficient. A security-aware scheduling mechanism, which plays an important role in QSCM, is capable of maximizing quality of security for real-time applications running in distributed systems as large-scale as Grids. Our empirical studies based on real world traces from a supercomputing center demonstratively show that the proposed model can significantly improve the performance of Grids in terms of both security and schedulability.

Streaming services and visual communication services delivered over the Internet have become popular in recent years. In the future, broadband services using MPEG2/4 will become the dominant type. These services will require transport protocols that provide high quality and high throughput from end to end of the system. We propose a new transfer method that allows the network load to be adaptively balanced according to the network's state. We built a prototype of an actual MPEG2 streaming system and used it to estimate the effectiveness of this method.

Emerging multimedia technologies introduce the prevalent multicast transmission, and the multicast tree is determined using the time-invariant network parameters. This paper addresses the time-varying multicast tree problem and presents path selection heuristics for multicast routing to determine an alternative path for real-time applications. A network is partitioned into the optimal region, the disjoint region, and the edge cutset if a branch of the multicast tree meets the un-guaranteed QoS condition. The path selection heuristics operate during the multicast session phase to efficiently select an alternative routing path containing an edge in the edge cutset to connect the multicast tree again. The source-based heuristics PS-SPT finds the path for minimal source-to-destination delay and the sharing-based heuristics PS-DDMC for minimal total cost. These path selection heuristics can efficiently provide solutions to keep the multicast transmission reliable. Simulation results also show that the proposed heuristics can provide effective good solutions for real-time multicast transmission. PS-SPT can select a path with optimal source-to-destination delay and PS-DDMC can select a path with optimal total cost.

Rate monotonic traffic scheduling algorithm had been shown to be the optimal static priority assignment scheme. The system model studied in can be considered (although not realistic) as a preemptive multiplexer which accepts constant bit rate connections that generate packets periodically. The multiplexer adopts a service discipline such that a lower priority packet can be preempted at any stage by a higher priority one without any loss. The constraint is that every packet has to complete its service before the arrival of its succeeding packet generated by the same connection. In this paper, we study the schedulability problem of rate monotonic schedulers for a fixed-length packet switched network such as the ATM network. A necessary and sufficient condition for a set of m constant bit rate connections to be rate monotonic schedulable is first derived and then utilized to design fast admission control algorithms. One of our algorithms computes in advance the minimum period of a connection which can be accepted given a set of existing connections.