An analytical paradigm to compare routing strategies in an ATMmultimedia environment

We propose an analytical model for the study of two different routing strategies in an ATM network supporting multimedia traffic flow: the multimedia virtual circuit (MVC) and the independent virtual circuit (IVC). The first strategy consists of assigning a single virtual channel to all of the monomedia streams which make up the multimedia flows; the second strategy consists of multiplexing homogeneous monomedia streams belonging to different multimedia sources on the same virtual channel. With this aim, a multimedia source is modeled as an arrival process defined as the superposition of heterogeneous correlated arrival processes, each of which models one monomedia source. In order to take into account the intermedia relationships which exist in a multimedia stream, each monomedia source is modeled as an interrupted Bernoulli process in which the transition and the arrival probabilities are functions of the states of the other monomedia sources. A finite-buffer discrete-time approach is used in order to compare MVC and IVC performance when an aggregate of N heterogeneous multimedia sources loads the network. Performance is evaluated for each monomedia source in terms of loss probability and jitter probability density function. To assess the proposed paradigm, a case study is shown     

A discrete-time paradigm to evaluate skew performance in amultimedia ATM multiplexer

One of the most important quality-of-service parameters in a multimedia environment is skew, defined as the difference between the delays suffered by the monomedia flows belonging to the same multimedia stream. An analytical paradigm is proposed to evaluate the skew affecting a multimedia traffic stream in an asynchronous transfer mode multiplexer. For this purpose, the emission process of each multimedia source loading the multiplexer is defined as the superposition of heterogeneous correlated emission processes, each of which models one monomedia source as a switched batch Bernoulli process (SBBP). In order to model the intermedia relationships, the transition probabilities in the Markov chain underlying each SBBP are functions of the state of the other monomedia sources. The model is applied to a case study and the dependence of skew performance on some of the source characteristics, such as intermedia correlation, and some of the environment characteristics, such as buffer size, output capacity, and buffer utilization is analyzed and discussed     

Discrete-time model for multimedia correlated sources

A multimedia source model is presented. To capture the intermedia synchronisation requirements of the streams in the multimedia flow, the model is defined as the superposition of heterogeneous correlated monomedia arrival processes. Transition probability matrices and correlation functions are calculated to allow any designer to investigate network performance by means of well-known analytical techniques     

Le terminal multimédia : de la bureautique à la communication

Learning and appropriation of new media by users and for the needs of corporations become real and concrete even if they are not yet well defined, stable or comprehensive. The multimedia concept asserts itself more and more, it should even be able due to high level of integration afforded by informatics to gather a great number of existing monomedia applications and to make them communicate on an all-purpose network. Analysis of what hinders this integration backs on observing how the potential user will behave towards each of the current media, graphics, full-motion video, sound and their evolving form. It is shown how the user must adapt to a new perception of this activity by means of handling virtual objects on a multimedia workstation.     

Multimedia service interworking over heterogeneous networkingenvironments

This article discusses technical issues related to delivery of multimedia services, such as video-on-demand, over heterogeneous networking environments. In particular, it describes the design and implementation of an experimental system for multimedia service interworking between a DAVIC (Digital Audio Visual Council) domain and an Internet domain. The system, called MIDDLEMEN (middlemen for multimedia environment), consists of the broker and the interworking unit, which work together to provide multimedia services across different types of networks in a seamless manner. The broker acts as a guide to multimedia services, and complements servers or clients that are not equipped with full functionality. The broker also controls the resources in MIDDLEMEN. The interworking unit performs various translation functions under the control of the broker, including stream transport protocol conversion, traffic monitoring and bit-rate control, service gateway conversion, and stream control conversion. Current implementation supports the delivery of a DAVIC-compliant VOD service from an ATM network to an IP network     

Optimal network selection in heterogeneous wireless multimedia networks

The complementary characteristics of different wireless networks make it attractive to integrate a wide range of radio access technologies. Most of previous work on integrating heterogeneous wireless networks concentrates on network layer quality of service (QoS), such as blocking probability and utilization, as design criteria. However, from a user’s point of view, application layer QoS, such as multimedia distortion, is an important issue. In this paper, we propose an optimal distributed network selection scheme in heterogeneous wireless networks considering multimedia application layer QoS. Specifically, we formulate the integrated network as a restless bandit system. With this stochastic optimization formulation, the optimal network selection policy is indexable, meaning that the network with the lowest index should be selected. The proposed scheme can be applicable to both tight coupling and loose coupling scenarios in the integration of heterogeneous wireless networks. Simulation results are presented to illustrate the performance of the proposed scheme.     

Calculation and analysis of compensation buffer size in multimediasystems

In multimedia systems end to end delay jitter has a great impact on the continuity of information playback. Therefore, it is necessary to compensate for delay variations to preserve the intramedia and intermedia temporal relationships. A buffer at the destination site is often adopted to compensate for the delay jitter. In this paper, a mapping relationship from P_QoS to buffer which is used to compensate for delay variations is given based on delay jitter statistics, and its theoretical analysis is also shown. So the buffer size of a multimedia synchronization system can be determined     

多媒体传感网中一种基于优化覆盖及压缩代价评估的节能机制

为降低无线多媒体传感网能耗,利用贪婪算法调整节点传感方向,实现优化覆盖并降低了传感开销.同时,对节点数据压缩代价进行评估,使其以最小的能量代价传输数据,并根据多媒体事件流规律,优化网络生存时间.仿真结果表明,本方法在延长网络生存时间方面表现出了较好性能.     

A synchronization framework for communication of pre-orchestratedmultimedia information

The authors propose a communication and presentation model called XOCPN, which can specify temporal relationships for the presentation of pre-orchestrated multimedia data, and communication requirements for the timely delivery of these data through multiple virtual circuits (MVCs). In particular, XOCPN contains connection set-up for guaranteeing required QoS parameters for efficient delivery of multimedia data, and various grains of stream synchronization (e.g., intra-stream synchronization, inter-object synchronization, and interstream pacing). The source and the destination interact in harmony and carry out different functions embedded in the transmitter and the receiver XOCPNs to accomplish synchronous playout of multimedia data. A protocol architecture and a set of service primitives have been proposed for the realization of the proposed model in a practical environment. Subsequently, the authors present a communication architecture that can be implemented using an ATM-based network, due to its flexibility in handling various types of traffic     

CASHeW: Cluster-based Adaptive Scheme for Multimedia Delivery in Heterogeneous Wireless Networks

Multimedia streaming over heterogeneous wireless networks has attracted significant interest in recent years from both telecom network operators and end users. However, the heterogeneity of the wireless network makes it very difficult to synchronize real-time multimedia streaming to different types of end-user devices across different wireless networks. In addition, with different delay and packet loss across different networks, multimedia delivery over the heterogeneous wireless networks cannot provide good quality streaming video. This paper proposes CASHeW—a novel cluster-based design with an in-built feedback-based adaptive mechanism that results in a higher video perceived quality in two-hop heterogeneous wireless network environments. CASHeW employs a proxy-client-server mechanism between the base station (BS) and the end-user; and importantly uses a quality-oriented adaptive scheme for efficient multimedia delivery. Simulation-based tests indicate that the performance of CASHeW not only outperforms transport layer adaptive delivery protocols like the TCP-Friendly Rate Control Protocol (TFRCP) and Loss Delay Adaptation (LDA+), but also is better than that of medium access control (MAC) layer protocols such as the Receiver Based Auto Rate (RBAR) and Enhanced Distributed Channel Access (EDCA) in terms of average perceived quality, average bit rate and loss rate.     

Video Streaming with Network Coding

Recent years have witnessed an explosive growth in multimedia streaming applications over the Internet. Notably, Content Delivery Networks (CDN) and Peer-to-Peer (P2P) networks have emerged as two effective paradigms for delivering multimedia contents over the Internet. One salient feature shared between these two networks is the inherent support for path diversity streaming where a receiver receives multiple streams simultaneously on different network paths as a result of having multiple senders. In this paper, we propose a network coding framework for efficient video streaming in CDNs and P2P networks in which, multiple servers/peers are employed to simultaneously stream a video to a single receiver. We show that network coding techniques can (a) eliminate the need for tight synchronization between the senders, (b) be integrated easily with TCP, and (c) reduce server’s storage in CDN settings. Importantly, we propose the Hierarchical Network Coding (HNC) technique to be used with scalable video bit stream to combat bandwidth fluctuation on the Internet. Simulations demonstrate that under certain scenarios, our proposed network coding techniques can result in bandwidth saving up to 60% over the traditional schemes.     

基于移动Agent平台的多媒体无线情景仿真

在无线多媒体通信系统中,视频信源在发送端按照一定的编码方式进行编码,形成视频帧,然后在网络层和传输层封装成IP数据包,经过无线信道传输到达接收端,在接收端进行解包和恢复,得到视频帧,再形成重建后的视频流。在此过程中,无线信道的传输特性会影响到视频的传输质量。文中在利用Agent平台的基础上,综合NS2仿真工具,建立研究多媒体信息在无线网络中的传输平台。仿真结果表明,由于无线信道特性的不理想,在接收端恢复视频信息与视频信源相比较,会出现一定的失真     

Multimedia services over digital subscriber lines

We study the problem of reliable and yet efficient multimedia communications over asymmetric digital subscriber line (ADSL) through joint consideration of compression/coding and channel transmission techniques. The transmission of digital multimedia data requires the existing systems to be augmented with functions that can handle not only ordinary (nonmultimedia) data. In addition, the high volume of multimedia data can be handled efficiently only if all available system resources are carefully optimized. We consider the special characteristics of ADSL channels to formulate optimization criteria. We present a system where the encoder consists of a layer coder that divides and compresses the source data into coded layers of multimedia data with different performance and quality-of-service requirements. The encoded bit streams are then transmitted over a noisy channel, where channel noise may distort the data. The decoder removes all the coding and compression applied in the encoder to obtain as close as possible the original data. It is conceivable that for such a system the loss of information occurs not only during the source compression but also in the channel transmission. The source coding scheme and bit stream arrangement have a direct impact on the channel distortion. Therefore, the joint consideration of source/channel coding design is ideal to reduce the overall information loss     

Delay sensitive scheduling schemes for heterogeneous QoS over wireless networks

Future wireless networks will support the growing demands of heterogeneous and delay sensitive applications. In this paper, a users' satisfaction factor (USF) is defined to quantify quality of service (QoS) for different types of services such as voice, data, and multimedia, as well as for different delay constraints. This USF not only predicts the final delivered QoS during transmission, but also take advantages of the fact that different packets can be decoded at different time in the receivers. Based on this USF, four types of scheduling schemes considering tradeoffs between system performance and individual fairness are proposed. These schemes explore the time, channel, and multi-user diversity to guarantee quality of service and enhance the network performance. From the simulation results, the proposed scheduling schemes achieve different tradeoffs between individual fairness and high system performance for the heterogeneous and delay sensitive applications, compared with the weighted round-robin and the modified proportional fairness scheduling schemes     

5G mobile terminals with advanced QoS-based user-centric aggregation (AQUA) for heterogeneous wireless and mobile networks

In this paper we present an advanced QoS provisioning module with vertical multi-homing framework for future fifth generation (5G) mobile terminals with radio network aggregation capability and traffic load sharing in heterogeneous mobile and wireless environments. The proposed 5G mobile terminal framework is leading to high performance utility networks with high QoS provisioning for any given multimedia service, higher bandwidth utilization and multi-RAT capabilities. It is using vertical multi-homing and virtual QoS routing algorithms within the mobile terminal, that is able to handle simultaneously multiple radio network connections via multiple wireless and mobile network interfaces. Our 5G proposal is user-centric, targeted to always-on connectivity, maximal network utilization, maximal throughput, seamless handovers and performances improvement by using vertical multi-homing, as well as session continuity. The performance of our proposed mobile terminal framework for 5G is evaluated using simulations and analysis with multimedia traffic in heterogeneous mobile and wireless scenarios with coexistence of multiple radio access technologies, such as 3G, 4G as well as future 5G radio access networks.     

An integrated adaptive bandwidth-management framework for QoS-sensitive multimedia cellular networks

Bandwidth is an extremely valuable and scarce resource in a wireless network. Therefore, efficient bandwidth management is necessary in order to provide high-quality service to users in a multimedia wireless/mobile network. In this paper, we propose new online bandwidth-management algorithms for bandwidth reservation, call admission, bandwidth migration, and call-preemption strategies. These techniques are combined in an integrated framework that is able to balance the traffic load among cells accommodating heterogeneous multimedia services while ensuring efficient bandwidth utilization. In addition, our online framework to adaptively control bandwidth is a cell-oriented approach that has low complexity, which makes it practical for real cellular networks. Simulation results indicate the superior performance of our bandwidth-management framework to strike the appropriate performance balance between contradictory quality-of-service requirements.     

Mobility of sources and listeners: real-time support of multicast services

In today??s Internet there is an increasing demand for group-based multimedia sessions, such as video/audio conferencing, IPTV, push media. Moreover, there is an increasing trend for mobile communications, with large network dynamic events, which require fast resilience to keep, as much as possible, sessions alive over time. In this sense, the bandwidth-constrained data transport scheme of IP multicast allows traffic optimization throughout the network. However, current IP multicast support over mobile communications is not efficient, mainly due to the IP addressing scheme and mobility. If multicast sources mobility is taken into account, then the performance of IP multicast is significantly degraded, due to the requirement for constantly changing the overall multicast tree. In this paper we propose a new solution for agent-based multicast in both sources and listeners mobile environments, called MUlticast TEleport (MUTE). MUTE considers the existence of anchor points in the network that provide proxy features, assuming the view of multicast data source inside the network, and establishing independence between listeners and sources movements. Moreover, we propose MUTE extension to efficiently support time-sensitive sessions, which consists in integrating IEEE 802.21 and QoS-aware abstracted transport approach, to allow media independent transport with fast resilience support. The evaluation of MUTE was carried out in NS-2, and the results show that it is able to provide both multicast sources and listeners mobility with decreased disruption time and increased network performance in heterogeneous systems.     

Routing Protocol for Heterogeneous Hierarchical Wireless Multimedia Sensor Networks

Recently, wireless multimedia sensor networks (WMSNs) have emerged as one of the most important technologies, driven by the development of powerful multimedia device such as CMOS. WMSNs require several factors such as resource constraints, specific QoS, high bandwidth and so on. In this paper, we propose a hierarchical heterogeneous network model based routing protocol for WMSNs. In our proposal, proposed network model is classified into monitoring class, delivery class and aggregation class. Also we define two kinds of the routing path in order to ensure bandwidth and QoS. In simulation results, we illustrate the performance of our proposal.