Time-Triggered Controller Area Network is widely accepted as a viable solution for real-time communication systems such as in-vehicle communications. However, although TTCAN has been designed to support both periodic and sporadic real-time messages, previous studies mostly focused on providing deterministic real-time guarantees for periodic messages while barely addressing the performance issue of sporadic messages. In this paper, we present an O(n2) scheduling algorithm that can minimize the maximum duration of exclusive windows occupied by periodic messages, thereby minimizing the worst-case scheduling delays experienced by sporadic messages.

The authors propose a user-centric seamless handover (HO) scheme, which is a kind of a vertical HO from a new perspective, toward a next generation network where heterogeneous access networks converge. The users' experience-oriented scheme allows users to enjoy the optimal service quality for real-time applications in respective access networks. In addition, the scheme sustains on-going sessions during the vertical HO. The proposed scheme consists of two methods -- the bicasting of Different Quality-level Streams (DiffQS) and the Delay Difference Absorption (DDA) method. Initially, the authors propose two plausible methods for the SIP-based bicasting of DiffQS. This document introduces a SIP-capable network element named the HO Assistive Server (HOAS). HOAS controls bicasting of DiffQS and provides users with the optimal service quality for real-time applications via respective access networks as well as avoiding packet loss during HO. The DDA method is also proposed to prevent a service interruption and smoothly continue a real-time service during HO. Evaluation results show that the scheme achieves the seamless service continuity from the users' perspective for HO between cellular and high-speed wireless access via the implementation of a prototype system.

A key issue in QoS-provisioning real-time wireless communications is to provide the QoS requirement with low energy consumption. In this paper, we propose an energy-efficient error correction scheme for real-time communications with QoS requirements in wireless networks. The QoS requirement of a message stream is modeled with (m, k) constraint, implying that at least m messages should be sent to a receiver during any window of k periods. The proposed scheme adaptively selects an error correcting code in an energy-efficient manner so that it maximizes the number of QoS provisionings per unit energy consumption.

In this paper, we propose a statistical method of time synchronization for computer clocks that have precisely frequency-synchronized oscillators. This method not only improves the accuracy of time synchronization but also prevents degradation in the frequency stability of the precise oscillators when the errors in the measured time offsets between computer clocks caused by network traffic possess a Gaussian distribution. Improved accuracy of time synchronization is achieved by estimating the confidence interval of the measured time offsets between the clocks. Degradation in frequency stability is prevented by eliminating unnecessary time correction for the computer clock, because time correction generally causes changes in the frequency accuracy and stability of the precise oscillators. To eliminate unnecessary time correction, our method uses an extended hypothesis test of the difference between the current mean and the mean at the last time adjustment to determine whether time correction is needed. Evaluation by simulating changes in the time offset of the existing ISDN clock synchronization system showed that this method achieves accurate time and stable frequency synchronization.

A variety of real-time multicast applications such as video conferences, remote lectures, and video-on-demand have become in commonplace with the expansion of broadband Internet services. Due to nontrivial problems in the IP multicast technology, the peer-to-peer multicast technology (P2P-multicast) has emerged as a practical implementation, although its network resource utilization is less efficient. A multihome network has the potential of alleviating this inefficiency by providing flexibility in communication path selections for each host with multiple gateways to the Internet. This paper has first formulated the P2P-multicast routing problem in the multihome network, and has proved the NP-completeness of its decision problem. Then, a two-stage heuristic algorithm called P2PMM_router has been presented for this P2P Multicast Multihome-network routing problem. The first stage constructs an initial multicast routing tree from an optimum spanning tree by Prim algorithm, through satisfying the constraints. The second stage improves the tree by repeating partial modifications and constraint satisfactions. The extensive simulation results using random network instances support the effectiveness of our P2PMM_router.

In future mobile networks, new technologies will be needed to enable a mobile host to move across heterogeneous wireless access networks without disruption of the connection. In the past, many researchers have studied handover in such IP networks. In almost all cases, special network devices are needed to maintain the host's mobility. Moreover, a host cannot move across heterogeneous wireless access networks without degradation of the goodput for real-time communication, although a mobile host with multiple network interfaces can connect to multiple wireless access networks. For these reasons, we consider that a mobile host needs to manage seamless handover on an end-to-end basis. In this paper, we propose a multi-path transmission algorithm for end-to-end seamless handover. The main purpose of this algorithm is to improve the goodput during handover by sending the same packets along multiple paths, minimizing unnecessary consumption of network resources. We evaluate our algorithm through simulations and show that a mobile host gains a better goodput.

Timed token protocols inadequately provide periodic communication service, although this is crucial for hard real time systems. We propose an algorithm to guaranteeing periodic communication service on a timed token protocol network. In this approach, we allocate bandwidth to each node so that the summation of bandwidth allocations is Target Token Rotation Time (TTRT). If a node cannot consume the allocated time, the residual time is made concession to other nodes for non-periodic service using a timer which contains the unused time value and is appended to the token. This algorithm can always guarantee transmission of real-time messages before their deadlines when the network utilization is less than 50%.

An approach to the enhancement of speech signals corrupted by additive colored noise is proposed and the system architecture to implement the proposed idea in real-time communication is introduced in this paper. A combination of a bandpass FIR filtering technique with wiener filtering is used to improve the SNR for speech signals. The average SNR improvement (between input and output SNR) is 22.48 dB. The additive noises are the sound from a turbo prop aircraft. The system, which shows excellent performance, is designed based on a 16 bits fixed point DSP (ADSP-2181) from Analog Devices. Experiment results demonstrate that the FIR filter leads to a significant gain in SNR, thus visibly improvement for the quality and the intelligibility of the speech.

Designing control and robotic systems as autonomous decentralized systems introduces a new degree of flexibility in the manufacturing and in the application of such systems. This flexibility is required for the systems to work in environments that are not totally predictable and that can change dynamically. In this paper, we present a new concept for real-time communication that supports this flexibility while still preserving real-time guarantees for hard real-time communication. The concept is designed to work on multiple-access busses. In particular, we consider its application on wireless local area networks and field-busses. The concept addresses requirements of hard-real time, soft real-time and non real-time communication. For this, we extend the TDMA (time- division multiple-access) approach for time-triggered hard-real time communication by the concept of shared channels that support event-triggered communication and coexist with hard real-time channels. A first implementation of concept has been carried out in the context of the CAN-bus.

Packet scheduling is one of the key mechanisms that will be employed in the network nodes (routers and switches) for supporting multiple quality of services. In this paper we propose a new packet scheduling algorithm called Urgency-based Round Robin (URR) which computes an index for flows in order to keep track of instantaneous bursts. Basically the index is employed as a measure of the time-dependent service necessity for each flow thus making it possible to detect those flows which might be in need of momentary service. Also, we propose a novel weight allocation scheme to be used together with the scheduler with the aim of preventing network underutilization. Our algorithm can be considered as a version of Weighted Round Robin (WRR) with improved delay characteristics. We show analytically that URR has the desired capability of upper-bounding unfairness. We also show, by simulation, that URR can improve delay performance even under extremely bursty traffic conditions without bandwidth overprovisioning. We also give simulation results for network traffic which exhibits long range dependency (self-similarity) and show that URR is again more effective than a plain round robin multiplexer.

On wide area networks (WANs), UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission. Meanwhile, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol (TCP) traffic. As opposed to UDP's simplicity, TCP adopts a unique flow control mechanism with sliding windows. Hence, the quality of service (QoS) of real-time applications using UDP is affected by TCP traffic and its flow control mechanism whenever TCP and UDP share a bottleneck node. In this paper, the characteristics of UDP packet loss are investigated through simulations of WANs conveying UDP and TCP traffic simultaneously. In particular, the effects of TCP flow control on the packet loss of real-time audio are examined to discover how real-time audio should be transmitted with the minimum packet loss, while it is competing with TCP traffic for the bandwidth. The result obtained was that UDP packet loss occurs more often and successively when the congestion windows of TCP connections are synchronized. Especially in this case, the best performance of real-time audio applications can be obtained when they send-small sized packets without reducing their transmission rates.

Distributed multimedia applications are often sensitive to the Quality of Service (QoS) provided by the communication network. They usually require guaranteed QoS service, so that real-time communication is possible. However, searching a route with multiple QoS constraints is known to be a NP-complete problem. In this paper, we propose a new simple and efficient distributed QoS routing algorithm, called "DQoSR," for supporting real-time communication in high-speed networks. It searches a route that could guarantee bandwidth, delay, and delay jitter requirements. Routing decision is based only on the modified cost, hop and delay vectors stored in the routing table at each node and its directly connected neighbors. Moreover, DQoSR is proved to construct loop-free routes. Its worst case message complexity is O(|V|2), where |V| is the number of nodes in the network. Thus DQoSR is fast and scales well to large networks. Finally, extensive simulations show that average rate of establishing successful connection of DQoSR is very near to optimum (the difference is less than 0.4%).