多媒体通信中的多点路由问题

多点通信是网络支持多媒体业务的关键技术之一。本文在不同优化选路准则下,结合当前应用背景介绍了基于最短路径和共享树的多点路由算法及其应用环境和性能分析,在此基础上对有关协议进行了讨论,最后结合多媒体业务的特征分析了多点路由算法的几个发展方向,以期对多点通信的路由问题最近和将来的研究给出必要的背景。     

多媒体通信的多播路由算法

在多媒体通信网的实际应用中,多播(multicasting)技术日显重要,在实际网络中,网络节点具备不同的多播能力,有些节点不具备多播能力,而具备多播能力的节点要限制其复制信息的数量,即节点多播能力受限,该文用节点的度约束来表示每个节点的多播能力;此外网络中的很多业务要求信息从源节点传送到目的节点的时延受限;因此该文研究带度约束和时延约束的多播路由问题,给出了一种Lagrange松弛法,能够较好地解决这类问题。     

多媒体通信的成组多点广播路由算法

多点广播是一源点传送信息到多个目的节点,成组多点广播是一组节点内部互相进行多点广播。成组多点广播的路由算法是为组内的每一个节点建立一棵路由树,用于点到多点的广播通信。本文提出一种新的成组广播路由算法,它比传统的成组广播路由算法在性能上有了一定的提高,同时更为简洁。     

Quality-of-service routing for supporting multimedia applications

Several new architectures have been developed for supporting multimedia applications such as digital video and audio. However, quality-of-service (QoS) routing is an important element that is still missing from these architectures. In this paper, we consider a number of issues in QoS routing. We first examine the basic problem of QoS routing, namely, finding a path that satisfies multiple constraints, and its implications on routing metric selection, and then present three path computation algorithms for source routing and for hop-by-hop routing     

Audio-to-visual conversion for multimedia communication

Although humans rely primarily on hearing to process speech, they can also extract a great deal of information with their eyes through lipreading. This skill becomes extremely important when the acoustic signal is degraded by noise. It would, therefore, be beneficial to find methods to reinforce acoustic speech with a synthesized visual signal for high noise environments. This paper addresses the interaction between acoustic speech and visible speech. Algorithms for converting audible speech into visible speech are examined, and applications which can utilize this conversion process are presented. Our results demonstrate that it is possible to animate a natural-looking talking head using acoustic speech as an input     

Multicast routing strategies for zoned VP-based ATM networks

In ATM networks, the concept of virtual path (VP) greatly simplifies cell processing in switches. The virtual channel connection (VCC) can be more quickly and efficiently established by good strategies of resource management. The method of constructing virtual path and the strategies of managing and allocating resources greatly affect the performance of the system operation. We propose a new architecture and the corresponding methods of constructing virtual paths; various methods and strategies, such as bandwidth control, rerouting, resource management, and fault recovery, are studied (Lee and Shie 2000). This paper focuses on multicast routing and analyzes some algorithms for this model. Simulation results show the good performance in bandwidth utilization, blocking probability, and loss probability     

Object-oriented communication structures for multimedia datatransport

The evolving multimedia applications generate requirements for complex transport capabilities, i.e., functional features, in the end-to-end communication system such as handling of heterogeneity among communicating terminals, supporting finer levels of user-specifiable quality of data transport service, and synchronization of various data streams for delivery at users in real time. Accordingly, the communication system may be viewed as extending the basic capabilities provided by the backbone network (e.g., bandwidth allocation) into a set of transport capabilities suitable for complex applications. This paper presents: (1) an object-oriented view of the user interface to the communication system with an elegant separation of data transport functionalities, and (2) an approach to the design of underlying transport protocols. The object-orientation decomposes an application-level data transport into a set of network channel objects, with each channel object handling a separate data stream. The object interactions are modeled using a “data-flow programming” style, which allows a richer set of protocols to implement the communication system and offers flexibility to accommodate complex and heterogeneous subscriber services/terminals. The “data-flow programming” method also allows a high degree of communication level parallelism among data transport through channels. The view of a multimedia communication system as a “parameterizable black-box”, as underscored in the object-oriented structuring, allows easier interworking of the communication system with existing networks and easier integration of multimedia transport into programming environments     

无线多媒体传感器网络中一种改进路由算法

随着网络负载增加,经典的TPGF( Two-Phase geographic Greedy Forwarding)算法难以找到节点分离路径,会导致网络吞吐量、投递率以及端到端时延性能下降。此外,当网络拓扑变动不大时, TPGF中每条路径所包含节点要消耗比其他节点更多的能量,会导致其过快死亡,从而影响网络性能。为此,将联合网络编码技术引入 TPGF,提出一种编码与能量感知的 TPGF 路由算法( NE-TPGF)。该算法综合考虑节点的地理位置、编码机会、剩余能量等因素,同时利用联合网络编码技术进一步扩展编码结构,充分利用网络编码优势来建立相对最优的传输路径。仿真结果表明, NE-TPGF能够增加编码机会,提高网络吞吐量和投递率,降低端到端时延,并且还有利于减少和平衡节点的能量消耗。     

Minimizing Intermediate Multicast routing for dynamic multi-hop ad hoc networks

A Minimizing Intermediate Multicast Routing protocol （MIMR） is proposed for dynamic multi-hop ad hoc networks. In MIMR, multicast sessions are created and released only by source nodes. In each multicast session process, the source node keeps a list of intermediate nodes and destinations, which is encapsulated into the packet header when the source node sends a multicast packet. Nodes receiving multicast packets decide to accept or forward the packet according to the list. Depending on topology matrix maintained by unicast routing, the shortest virtual hierarchy routing tree is constructed by improved Dijkstra algorithm. MIMR can achieve the minimum number of intermediate nodes, which are computed through the tree. No control packet is transmitted in the process of multicast session. Load of the network is largely decreased. Experimental result shows that MIMR is flexible and robust for dynamic ad hoc networks.     

Cross-layer multipath routing scheme for wireless multimedia sensor network

Wireless Networks - Multimedia transmission in wireless multimedia sensor network requires restricted quality of services (QoS) conditions. Where the resource-constrained nature of wireless...     

基于灰狼算法的无线网络组播路由优化

针对无线网络中资源受限的组播路由问题,考虑网络节点的节点度限制和网络链路的带宽约束,以最小化组播路由开销为目标,提出了一种二进制编码方式的基于灰狼优化算法的组播路由策略.在给定的网络拓扑下,基于灰狼优化算法的组播路由策略可以迅速找到一棵包含源和目的节点的最小开销组播树.仿真结果表明,相比于遗传算法,所提出的基于灰狼优化...     

Interoperable adaptive multimedia communication

The desire to gain access to rich, multimedia-based information anywhere, anytime grows enormously. To achieve such access, the research and standardization communities have launched an initiative called universal multimedia access (UMA). However, UMA tools and specifications that have so far emerged concentrate mostly on constraints imposed by terminals and networks along the multimedia delivery chain; users who consume the content are rarely considered. Researchers have developed a plethora of technologies and standards to address some of the issues. However, the big picture of how these different technologies and standards fit together is missing. Thus, the moving picture experts group (MPEG) decided to standardize the MPEG-21 multimedia framework with the ultimate goal to support users during the exchange, access, consumption, trade, or other manipulation of so-called digital items in an efficient, transparent, and interoperable way. Device and coding-format-independent multimedia content adaptation standardization committees such as MPEG, the Internet Engineering Task Force (IETF), and the World Wide Web Consortium (W3C) are exploring device and coding format independence issues. Here, however, we focus on how we can use the tools specified within MPEG-21 for interoperable multimedia communication.     

Low bit-rate speech coders for multimedia communication

The International Telecommunications Union (ITU) has standardized three speech coders which are applicable to low-bit-rate multimedia communications. ITU Rec. G.729 8 kb/s CS-ACELP has a 15 ms algorithmic codec delay and provides network-quality speech. It was originally designed for wireless applications, but is applicable to multimedia communications as well. Annex A of Rec. G.729 is a reduced-complexity version of the CS-ACELP coder. It was designed explicitly for simultaneous voice and data applications that are prevalent in low-bit-rate multimedia communications. These two coders use the same bitstream format and can interoperate. The ITU Rec. G.723.1 6.3 and 5.3 kb/s speech coder for multimedia communications was designed originally for low-bit-rate videophones. Its frame size of 30 ms and one-way algorithmic codec delay of 37.5 ms allow for a further reduction in bit rate compared to the G.729 coder. In applications where low delay is important, the delay of G.723.1 may be too large. However, if the delay is acceptable, G.723.1 provides a lower-complexity alternative to G.729 at the expense of a slight degradation in quality. This article describes the attributes of speech coders such as bit rate, complexity, delay, and quality. Then it discusses the basic concepts of the three new ITU coders by comparing their specific attributes. The second part of this article describes the standardization process for each of these coders     

Millimeter waves for short-range multimedia communication systems

A framework for the performance characterization of short-range communications systems is developed with the intention of investigating the feasibility of new multimedia wireless services at millimeter waves (MMWs). Both narrow- and wide-band systems are considered for mobile and/or fixed users. This paper aims at defining and evaluating proper metrics to characterize the service quality for the user and jointly takes the propagation characteristics, the transmission techniques, and the multiple access protocols into account. The definition of service-oriented metrics is emphasized. Three different real scenarios operating at MMW are investigated with a unified perspective: intelligent transport systems, wide-band local-area networks, and local multipoint distribution systems for interactive video services. The role played by the MMW band in the development of these services is discussed. In each scenario, accurate propagation analysis is carried out and suitable countermeasure techniques are pointed out in order to join suitable service-quality levels. The methodology considered is based on both analytical and semianalytical tools for performance evaluation     

Resource efficient LDPC decoders for multimedia communication

Achieving high image quality is an important aspect in an increasing number of wireless multimedia applications. These applications require resource efficient error correction hardware to detect and correct errors introduced by the communication channel. This paper presents an innovative flexible architecture for error correction using Low-Density Parity-Check (LDPC) codes. The proposed partially-parallel decoder architecture utilizes a novel code construction technique based on multi-level Hierarchical Quasi-Cyclic (HQC) matrix. The proposed architecture is resource efficient, provides scalable throughput and requires substantially less power compared to other decoders reported to date. The proposed decoder has been implemented on a Xilinx FPGA suitable for WiMAX application and achieves a throughput of 548 Mbps. Performance evaluation of the decoder has been carried out by transmitting JPEG images over a wireless noisy channel and comparing the quality of the reconstructed images with those from other similar decoders.     

An architecture for real-time multimedia communication systems

A multimedia communication system includes both the communication protocols used to transport the real-time data and the distributed computing system (DCS) within which any applications using the protocols must execute. The architecture presented attempts to integrate these communications protocols with the DCS in a smooth fashion in order to ease the writing of multimedia applications. Two issues are identified as being essential to the success of this integration: the synchronization of related real-time data streams, and the management of heterogeneous multimedia hardware. The synchronization problem is tackled by defining explicit synchronization properties at the presentation level and by providing control and synchronization operations within the DCS which operate in terms of these properties. The heterogeneity problems are addressed by separating the data transport semantics (protocols themselves) from the control semantics (protocol interfaces)     

Multicast routing and bandwidth dimensioning in overlay networks

Multicast services can be provided either as a basic network service or as an application-layer service. Higher level multicast implementations often provide more sophisticated features and can provide multicast services at places where no network layer support is available. Overlay multicast networks offer an intermediate option, potentially combining the flexibility and advanced features of application layer multicast with the greater efficiency of network layer multicast. In this paper, we introduce the multicast routing problem specific to the overlay network environment and the related capacity assignment problem for overlay network planning. Our main contributions are the design of several routing algorithms that optimize the end-to-end delay and the interface bandwidth usage at the multicast service nodes within the overlay network. The interface bandwidth is typically a key resource for an overlay network provider, and needs to be carefully managed in order to maximize the number of users that can be served. Through simulations, we evaluate the performance of these algorithms under various traffic conditions and on various network topologies. The results show that our approach is cost-effective and robust under traffic variations.     

Delay constraint multipath routing for wireless multimedia ad hoc networks

According to the disadvantages of real time and continuity for multimedia services in ad hoc networks, a delay constraint multipath routing protocol for wireless multimedia ad hoc networks, which can satisfy quality of service (QoS) requirement (QoS multipath optimized link state routing [MOLSR]), is proposed. The protocol firstly detects and analyzes the link delay among the nodes and collects the delay information as the routing metric by HELLO message and topology control message. Then, through using the improved multipath Dijkstra algorithm for path selection, the protocol can gain the minimum delay path from the source node to the other nodes. Finally, when the route is launched, several node‐disjoint or link‐disjoint multipaths will be built through the route computation. The simulation and test results show that QoS‐MOLSR is suitable for large and dense networks with heavy traffic. It can improve the real time and reliability for multimedia transmission in wireless multimedia ad hoc networks. The average end‐to‐end delay of QoS‐MOLSR is four times less than the optimized link state routing. Copyright © 2014 John Wiley & Sons, Ltd.     

移动通信中的组播路由策略

终端移动或加入、退出组播组时，路由生成树应随之动态变化。对比多种组播路由策略后，在IntServ模型中融入CSM协议，希望在不增加网络负荷的前提下降低终端越区切换带来数据报的收发中断时间。     

Multicast routing with end-to-end delay and delay variationconstraints

We study the problem or constructing multicast trees to meet the quality of service requirements of real-time interactive applications operating in high-speed packet-switched environments. In particular, we assume that multicast communication depends on: (1) bounded delay along the paths from the source to each destination and (2) bounded variation among the delays along these paths. We first establish that the problem of determining such a constrained tree is NP-complete. We then present a heuristic that demonstrates good average case behavior in terms of the maximum interdestination delay variation. The heuristic achieves its best performance under conditions typical of multicast scenarios in high speed networks. We also show that it is possible to dynamically reorganize the initial tree in response to changes in the destination set, in a way that is minimally disruptive to the multicast session