Today, many audiovisual delivery systems, including video streaming and video conferencing, are being developed for use over a range of networking technologies, the differing characteristics of which pose problems for service level interoperability. Multimedia transcoding is one means to provide interoperability between different types of audiovisual terminals and between terminals that connect to different networks. In this paper, we will present a multimedia transcoder system, which provides interoperability between video conferencing terminals on IP networks and mobile terminals on mobile networks.

We propose and analyze a new efficient handoff scheme called Splitted-Rating Channel Scheme in UMTS networks, and we analyze the call level performance of splitted-rating channel scheme and then packet level performance of downlink traffic at UMTS circuit-switched networks. In order to reduce the blocking probability of originating calls and the forced termination probability of handoff calls, a splitted-rating channel scheme is applied to the multimedia UMTS networks. This multimedia network supports two classes of calls; narrowband call requiring one channel and wideband call requiring multiple channels. The channels in service for wideband call are splitted its channels for lending to originating call and handoff call according to threshold control policy. By assuming that arrivals of narrowband calls and arrivals of wideband calls are Poisson, we model the number of narrowband calls and the number of wideband calls in the one cell by Level Dependent Quasi-Birth-Death (QBD) process and obtain their joint stationary distribution. For packet level analysis, we first describe the downlink traffic from the base station to a mobile terminal in UMTS networks, and calculate the mean packet delay of a connected wideband call by using QBD analysis. Numerical examples show that our splitted-rating channel scheme reduces the blocking probability of originating call and the forced termination probability of handoff call with a little degradation of packet delay.

In this paper, we implemented practical resource adaptation mechanism for broadband application environment especially suitable for portable computers (note PCs) with limited power supply. Continuous development of portability enhancement and increasing computation with less power consumption for battery operating environment in note PC equips with intelligent resource controlling mechanism embedded inside the hardware. In such environment, where occasional environmental changes take in place rapidly, adaptive and collaborative controls of resources are required. Optimization of resource based on power supply management is dedicated to the operating environment. Bi-directional, user, application, operating system and device aware interface for resource configuration/management is extended to the current Unix operating system by implementing application programming interface of Advanced Configuration and Power Interface (ACPI). Traditional operating system conceals the active device probing and controlling method autonomously. Implementation of "State handler daemon" interface enables applications and application users to monitor and adapt the resources available during the operation. Thick collaboration between application and devices that are reserved in its limited environments economizes the consuming device utilization resulting: for example, life extension of battery life and effective network bandwidth adaptation. We focused on DVTS (Digital Video Transport System) as a typical broadband application. Evaluation shows 125 minutes of continuous battery operation with battery life extension compared to 70 minutes with traditional systems. Collaboration with network device management enables 100% packet transfer compared to 89% without any Resource adaptations.

The potential network architecture of the emerging carrier class VoIP (Voice over IP) technology for NGN (Next Generation Networks) adopts distributed control architecture to take full advantage of scalability, reliability, flexibility, and interoperability. However, the design of distributed control architecture in the carrier class VoIP network is the state-of-the-art in decentralization and distribution of control. Different configurations of system elements, control scheme of inter system elements communications, signaling protocol, functional partitioning, and scheduling of jobs in call control processing may affect the system performance and QoS (Quality of Service) of MGC (Media Gateway Controller) in carrier class VoIP network. Hence, the modeling of distributed control architecture and its performance analysis are essential issues whenever optimum control architecture has to be determined to meet design requirements. Based on these reasons, this paper proposes several potential network architectures and focuses on the performance study of distributed control architecture in carrier class VoIP network. The SIGTRAN-based distributed control architecture model and the MGCP/MEGACO-based distributed control architecture model are presented. Then, we analyze the SIGTRAN-based distributed control architecture model between MGC and SG (Signaling Gateway) using WRR (Weighted Round Robin) and WF2Q (Worst-case Fair Weighted Fair Queueing) scheduling algorithms respectively. And, we analyze the MGCP/MEGACO-based distributed control architecture model between MGC and MG (Media Gateway) using M/G/1 gating service queueing model. Consequently, the results of performance analysis can be used to evaluate whether the performance of distributed control architecture model can meet the requirement of planning and design for carrier class VoIP network deployment.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.

Future wireless communication systems will provide mobile terminals with high bit-rate transmissions for accessing broadband wired networks. In this paper, we envisage a Time Division Multiple Access - Time Division Duplexing (TDMA-TDD) air interface and we propose a Medium Access Control (MAC) protocol, named Dynamic Scheduling - TDD (DS-TDD), that allows guaranteeing the QoS of different traffic classes and efficiently supports uplink/downlink traffic asymmetries. The DS-TDD performance is theoretically analyzed. Moreover, the DS-TDD protocol is compared with another scheme proposed in the literature. Finally, the impact of packet errors on the DS-TDD performance is evaluated.

A hybrid data transmission technique for multimedia satellite broadcasting is proposed. The main-channel data and sub-channel data are simultaneously transmitted using QPSK modulation and 2ASK modulation, respectively, but the latter modulation timing is offset by half the main-channel QPSK symbol length in time. The BER performance in a Gaussian channel, the transmission bandwidth, and the transmit power peak factor are theoretically analyzed for various impulse responses of the sub-channel transmit filter. It is found that the use of the sub-channel transmit filter having a sine impulse response minimizes the sub-channel BER while keeping the transmission bandwidth and the transmit power peak factor lower than those of CAPSK transmission.

To diffuse multimedia information services, communication networks must guarantee the quality of services (QoSs) requested by users. In addition, users should be allowed to observe the network in order to customize their own services. A new network management architecture is therefore essential. It must perceive not only node connectivity, but also network failure points and the traffic situation dynamically. This paper introduces the network map as such an architecture on personalized multimedia communication networks and proposes multiple QoS routing using the network map. Moreover, a prototype system is built in order to verify the availability of the network map.

We propose a DPC-PA (Distributed Power Control with Power Assignment) algorithm to speed up the process of finding a feasible power set for multimedia CDMA wireless networks. We prove that given a feasible configuration, the power set of the DPC-PA algorithm will converge to a feasible power set which achieves equality in a set of linear equations. We also found, from numerical experiments, that the DPC-PA algorithm can find a feasible power set (if there is one) much faster than the distributed power control algorithm without power assignment.

In this paper, we propose an RTP/UDP/IP header compression method, Multiple-Reference Compression (MRC), which is designed for mobile multimedia communications. MRC is a compression method that calculates differences from the multiple reference headers that have already been sent and inserts them into a compressed header. The receiver can decompress the compressed header as long as at least one of the reference headers is correctly received and decompressed. MRC improves robustness against packet losses compared with CRTP defined in IETF RFC2508, and imposes less overheads and computational burden than robust header compression (ROHC) defined in RFC3095. We also implemented MRC and other header compression algorithms into our mobile testbed, and conducted multimedia streaming experiments over the testbed. The results of the experiments show that MRC offers the same level of packet loss rate as Legacy RTP for both audio and video streams, and provides better media quality than Legacy RTP and CRTP on error-prone radio links. Header compression robust against packet losses is expected as a key technology for VoIP and multimedia streaming services over 3G and future mobile networks.

The soft handoff is widely adopted in code division multiple access (CDMA) systems for its many advantages mainly resulting from site diversity. However, in the forward link, other cell interference can be increased by soft handoff, decreasing system capacity. In future mobile systems, provision for the sufficient forward link capacity is very important since the forward link load is much higher than the reverse link load in mobile multimedia services such as Internet access. In this paper, we consider a combined handoff strategy in which voice services are provided with soft handoff whereas data services are supported with hard handoff. We analyze the effect of handoff method on the forward link performance. The performance measures we use are the outage probability of the bit energy to noise density ratio and the capacity based on the outage probability. As a result, we show that the combined handoff is very useful in CDMA cellular networks supporting both voice and data services simultaneously.

A generic multilevel quality-of-service (QoS) model for distributed multimedia applications is presented. QoS mapping mechanisms are required to translate the QoS parameters among the hierarchical levels. One QoS mapping mechanism based on the spline functions is proposed, hence two splines are compared. One is natural splines and the other is B-splines. QoS measurement experiments were conducted, and it is found that the B-splines give more accurate mapping results than the natural splines once the knots for the splines are selected appropriately.

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%.

This paper proposes a scheme for joint synchronization between stored media with interactive control and live media in multicast communications. We deal with visual search control, such as fast-forward and fast-reverse, as interactive control. The proposed scheme enables visual search by enhancing the virtual-time rendering (VTR) media synchronization algorithm, which the authors previously proposed, and adjusts the timing of changing the visual search mode among destinations by carrying out group synchronization control. We also demonstrate the effectiveness of the scheme by experiment.

Scalabilities of bit rate and coding format in coded multimedia contents have become very important for the efficient use of network bandwidth and storage capacity with the recent availability of a wide variety of bandwidth and storage media. However, the conventional approach uses decompression and recompression processes to realize the above scalabilities, which require very expensive computations. In addition, a very large cache space is required for storing the decoded audio-video data. This paper describes three fast scalability methods for MPEG audio and video data, MPEG audio/video bit rate conversion and MPEG format conversion, in order to address these problems. As for the first scalability, MPEG audio coding bit rate conversions, we describe subband domain conversion using bandwidth limitation, requantization and a requantization reflecting phychoacoustic model. Four types of MPEG video bit rate conversion are described that use bandwidth limitation, out-loop requantization, in-loop requantization, and hybrid requantization. As for the format conversion, the fast baseband domain format conversion is performed using coding information such as motion vectors and coding types extracted from input coded video. The experimental results of several comparisons with the above scalabilities and conventional transcoding methods are also shown.

A novel optical transmission line consisted of fibers characterized by positive and negative medial dispersion of NZ-DSF and SMF was designed and fabricated. Both P-MDF and N-MDF have achieved the medial dispersion and low non-linearity simultaneously. Total characteristics were confirmed to be suitable for the future high-bit-rate transmission.

Media processing has become one of the dominant computing workloads. In this context, SIMD instructions have been introduced in current processors to raise performance, often the main goal of microprocessor designers. Today, however, designers have become concerned with the power consumption, and in some cases low power is the main design goal (laptops). In this paper, we show that SIMD ISA extensions on a superscalar processor can be one solution to reduce power consumption and keeping a high performance level. We reduce the average power consumption by decreasing the number of instructions, the number of cache references, and using dynamic power management to transform the speedup in performance in power consumption reduction.

A 4GOPS 3 way-VLIW image recognition processor for an automobile system has been developed. The processor is based on a configurable and extensible media processor enabling optimization for a specific application by means of design-time configuration. Using VLIW coprocessor extension, the processor can satisfy the performance requirements of the system. Overhead by VLIW-mode instructions is only 7%. The VLIW co-processor occupies only 12% of the die area. Thus, good cost-performance for media processing in each embedded system can be achieved by this configurable media processor.

Sophisticated and robust resource management is an essential issue in future wireless systems which will provide a variety of application services. In this paper, we employ an adaptive-network-based fuzzy inference system (ANFIS) to control the resource allocation for mobile multimedia networks. ANFIS, possessing the advantages of expert knowledge of fuzzy logic system and learning capability of neural networks, can provide a systematic approach to finding appropriate parameters for the Sugeno fuzzy model. The fuzzy resource allocation controller (FRAC) is designed in a two-layer architecture and selects properly the capacity requirement of new call request, the capacity reservation for future handoffs, and the air interface performance as input linguistic variables. Therefore, the statistical multiplexing gain of mobile multimedia networks can be maximized in the FRAC. Simulation results indicate that the proposed FRAC can keep the handoff call blocking rate low without jeopardizing the new call blocking rate. Also, the FRAC can indeed guarantee quality of service (QoS) contracts and achieve higher system performance according to network dynamics, compared with the guard channel scheme and ExpectedMax strategy.

An on-chip, 64-Mb, embedded, DRAM MPEG-2 encoder LSI with a multimedia processor has been developed. To implement this large-scale and high-speed LSI, we have developed the hierarchical skew control of multi-clocks, with timing verification, in which cross-talk noise is considered, and simple measures taken against the IR drop in the power lines through decoupling capacitors. As a result, the target performance of 263 MHz at 1.5 V has been successfully attained and verified, the cross-talk noise has been considered, and, in addition, it has become possible to restrain the IR drop to 166 mV in the 162 MHz operation block.