In this letter, we propose a Hadoop-based Video Transcoding System (HVTS), which is designed to run on all major cloud computing services. HVTS is highly adapted to the structure and policies of Hadoop, thus it has additional capacities for transcoding, task distribution, load balancing, and content replication and distribution. To evaluate, our proposed system, we carry out two performance tests on our local testbed, transcoding and robustness to data node and task failures. The results confirmed that our system delivers satisfactory performance in facilitating seamless streaming services in cloud computing environments.

Information-Centric Networking (ICN) technology has recently been attracting substantial interest in the research community as one of the most promising future Internet architectures. The Named Data Networking (NDN) approach, which is one of the most recent instantiations of the ICN approach, would be a good choice for multimedia services, because NDN utilizes in-network storage embedded in NDN routers by caching recently or frequently requested contents. It is important to determine which data to cache at which NDN routers in order to achieve high performance, by considering not only the popularity of contents but also the inter-chunk popularity of a content item. This paper presents a chunk-block-based incremental caching scheme that considers both content and inter-chunk popularity. Our proposed scheme employs an incremental cache populating mechanism, which utilizes not only core-side but also edge-side NDN routers according to the request rate of the content item. Through simulations, we show that the proposed scheme achieves less delay, reduced redundant network traffic, and a higher cache hit ratio than legacy schemes.

Network intelligence is a discipline that builds on the capabilities of network systems to act intelligently by the usage of network resources for delivering high-quality services in a changing environment. Wide area network intelligence is a class of network intelligence in wide area network which covers the core and the edge of Internet. In this paper, we propose a system based on machine learning for wide area network intelligence. The whole system consists of a core machine for pre-training and many terminal machines to accomplish faster responses. Each machine is one of dual-hemisphere models which are made of left and right hemispheres. The left hemisphere is used to improve latency by terminal response and the right hemisphere is used to improve communication by data generation. In an application on multimedia service, the proposed model is superior to the latest deep feed forward neural network in the data center with respect to the accuracy, latency and communication. Evaluation shows scalable improvement with regard to the number of terminal machines. Evaluation also shows the cost of improvement is longer learning time.

In the infrastructure-less disaster environment, the application of the peer-to-peer (P2P) group conference over mobile ad hoc network (MANET) can be used to communicate with each other when the rescue crews search the survivors but work separately. However, there still are several problems of in-time multimedia delivery in P2P-MANET: (1) MANET mobility influences the maintenance of P2P overlay. (2) P2P overlay is not proximal to MANET topology, this leads to the inefficient streaming delivery. (3) The unreliable wireless connection leads to the difficulty of multi-source P2P group conferencing. Therefore, P2P conferencing cannot work well on MANET. To overcome the above disadvantages, in this paper, we present a cross-layer P2P group conferencing mechanism over MANET, called RING (Real-time Intercommunication Network Gossip). The RING uses the ring overlay to manage peers and utilizes the cross-layer mechanism to force the ring overlay to be proximal to MANET topology. Therefore, RING can lead efficient in-time multimedia streaming delivery. On the other hand, the ring overlay can deal with peer joining/leaving fast and simply, and improves the delivery efficiency with the minimum signaling overhead. Through mathematical theory and a series of experiments, we demonstrate that RING is workable and it can shorten the source-to-end delay with minimal signaling overhead.

Bandwidth is an extremely valuable and scarce resource in multimedia networks. Therefore, efficient bandwidth management is necessary in order to provide high Quality of Service (QoS) to users. In this paper, a new QoS-aware bandwidth allocation algorithm is proposed for the efficient use of available bandwidth. By using the multi-objective optimization technique and Talmud allocation rule, the bandwidth is adaptively controlled to maximize network efficiency while ensuring QoS provisioning. In addition, we adopt the online feedback strategy to dynamically respond to current network conditions. With a simulation study, we demonstrate that the proposed algorithm can adaptively approximate an optimized solution under widely diverse traffic load intensities.

Future optical code division multiple access (CDMA) networks should be designed for multirate and fully integrated multimedia services. In the conventional schemes, multilength optical orthogonal codes (OOCs) are designed to support multirate systems, while variable-weight OOCs are designed to support differentiated quality of service (QoS) for multimedia applications. In this paper, a novel class of optical signature codes; multiple-length variable-weight optical orthogonal codes (MLVW-OOC) is proposed for supporting multirate and integrated multimedia services in optical CDMA networks. The proposed MLVW-OOC has features that are easy to construct variable-weight codes and expanded to multiple-length codes. A construction method for designing MLVW-OOCs up to three levels of codes is discussed. The designed MLVW-OOCs can support differentiated requirements on data rates and QoS for several types of services in the networks. A code analysis for obtaining the value of cross-correlation constraints or multiple access interference (MAI) computation for several levels of codes is also suggested. The cross-correlation constraints of the proposed codes are better than the conventional codes such as multilength OOCs. Finally, the bit error probability performance of the two-level MLVW-OOC is evaluated analytically. The results show that the proposed MLVW-OOC can provide differentiated bit error probability performances for several combinations of data rates and QoS.

Scheduling algorithms are playing a key role in overall system performance of broadband wireless systems (BWS). Maximal SNR (MaxSNR) and Round Robin (RR) are two conventional scheduling strategies which emphasize efficiency and fairness respectively. Proportional Fair (PF) algorithm provides tradeoff between efficiency and fairness. In this paper, we apply PF to IEEE 802.16a OFDM based BWS and name it OPF. We also propose a new algorithm for multimedia services: Normalized Multimedia Adaptive OPF (NMAOPF). Adaptive modulation and coding scheme is applied in time varying and frequency selective fading wireless channel. System performances are compared in efficiency and fairness with and without user mobility. Efficiency is in terms of throughput, mean packet delay and packet drop ratio; fairness is in terms of user satisfaction rate and average user rate. Joint PHY and MAC layer simulation results show that: within the traffic range of 55 to 70 Mbps, compared with RR and MaxSNR, the performance of OPF is in between. Our proposed NMAOPF outperforms all others without user mobility, while under mobility, it is not as good as MaxSNR but better than OPF and RR.

Takine investigated the continuous time queueing system with single server and Markovian arrival streams (MAS) where service times of customers depend on the states of the underlying Markov chain immediately before and after arrivals and may differ from one another among streams. In this paper, we consider the discrete version of Takine's model. Since a MAS can be used to model the correlated traffics of several classes, we consider the discrete time queueing system with MAS as an input traffic. Moreover packets from different classes of traffic may require different service times. So service times of customers which are different from one another among streams and may depend on the states of the underlying Markov chain immediately before and after arrivals, are also considered. We obtain the distribution of the workload in the system using Neuts' matrix analytic method. Based on this result, we obtain the distributions of the actual waiting time and the sojourn time of a customer of each class and the joint distribution of the numbers of customers of each class. We also provide a recursion formula for computing the joint queue length distribution. We give an application to multimedia conference system and numerical example.

In this paper, we present an Internet personal live-broadcasting server system. Our solution is not only for experts but also for amateur users who want to broadcast using simple multimedia equipment. For scalable broadcasting services, we developed multiple-channel establishment and channel expansion. Concurrent services for a large number of broadcasting channels are effectively provided. Also, the capacity of channels can be expanded as the number of participants increases. Furthermore, for the sake of complete live broadcasting with high-quality transmission, the system supports both TCP (transmission control protocol) and UDP (user datagram protocol) according to the status of network environments as well as the received packet loss in the user system. The performance of the system is effectively evaluated at such practical commercial sites as well-known community and E-business sites.

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.

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.

This paper proposes three MAC protocols over APON to provide residential and small business customers with multimedia services. The proposed protocols support the frame structure of ITU-T recommendation G.983.1 and also provide diverse ATM service classes such as CBR, rtVBR, nrtVBR, ABR, and UBR traffics. Each service is allocated on the basis of priority. Especially, for allocating CBR and rtVBR services, each protocol uses different cell arrival timing information which is achieved with specific coding and ranging procedure. Focusing the difference of cell arrival timing information, we will investigate the performance of proposed protocols. For the proposed MAC protocols, we present grant field format, minislot format, and bandwidth allocation algorithm. Computer simulation shows the performance of the proposed protocols in terms of CDV and delay, comparing with the normal FIFO protocol.

This paper proposes a novel MAC scheme over APON based on the cell arrival timing information to provide residential and small business customers with multimedia services. The proposed scheme supports the frame format of ITU-T recommendation G.983 and also provides diverse ATM service classes such as CBR, rtVBR, nrtVBR, ABR, and UBR traffics. Each service is allocated on the basis of priority and cell arrival timing information. Especially, the CBR and rtVBR services, which are sensitive to delay and CDV, are allocated with higher priority and more exact arrival timing resolution which is achieved with specific coding and ranging procedure. For the proposed MAC scheme, we present grant field format, minislot format, and bandwidth allocation algorithm. Computer simulation results show that the performance of the proposed scheme is significantly improved in terms of CDV and delay time in case of CBR and rtVBR services, comparing with the normal FIFO scheme.

Given the recent advances in telecommunication services, new service operations are needed that support not only existing network services, but also information and communication providing services, such as multimedia services. It is particularly important that service usage information should be gathered, processed, analyzed and presented properly for smooth service provisioning and business support. This paper defines a reference model for service usage information and an implementation model for establishing the functions needed to gather and process this information. The results of an service operation system as implemented in NTT multimedia service trials, currently underway in the city of Urayasu, are also described.

A leaky-bucket-with-gate algorithm is proposed to control connection-setup congestion in telecommunication networks providing multimedia services, in place of the call-gapping algorithm used in telephone networks. Multimedia services may use more than one connection simultaneously, while standard telephone services use only one connection at a time. A set of connections used to construct a multimedia service is called a correlated connection group, and the setup requests of such a group form correlated request group. A correlated request group is assumed to be accepted into the network only when all the connection-setup requests for the group are accepted. In this paper, the proposed leaky-bucket-with-gate algorithm, a pure leaky-bucket algorithm, and a call-gapping algorithm are evaluated by simulating traffic with a mix of correlated and uncorrelated connection-setup requests, which models setup requests for video conferencing and telephone services. The simulation results show that the proposed algorithm accepts correlated request groups more efficiently than the pure leaky-bucket and call-gapping algorithms under the simulated traffic conditions, except when the interarrival time in a correlated request group is longer than the acceptance interval. We also present queueing analysis for determining the control parameters in the proposed algorithm. Implementation of this algorithm will facilitate the handling of both setup request traffic for correlated connection groups and for uncorrelated connections in multimedia networks.

Development of a large-scale mobile communications network (IMN: Intelligent Mobile communications Network), as an infrastructure integrating multimedia functions, is indispensable for the support of future mobile communication services aiming toward "personalization," "intelligence," and "multimedia services." This paper discusses the aims of mobile communications and the outline of network technology aspects of PDC (Personal Digital Cellular) network which is currently in service. In addition, the future prospect of mobile communication technologies is discussed with special focuses on the support of universal mobility, network architecture including mobile communications platform, and multimedia technologies in the transport and access systems.

Broadband multimedia information environments are part of the next big advance in communications and computer technology. The use of multimedia infrastructures in offices is becoming very important. This paper deals with a service concept and human interfaces based on a paper metaphor. The proposed service offers the advantages of paper and eliminates the disadvantages. The power of multimedia's expressiveness, user interaction, and hypermedia technology are key points of our solution. We propose a system configuration for implementing the service/human interface.