Signaling challenges in video dial tone networks

Residential broadband networks present new signaling challenges as compared to the more traditional telephony networks. Access signaling protocols such as DSM-CC and Q.2931 for ATM, are evolving to meet these challenges. A desirable outcome will be a solution that supports seamless interworking among the newer video applications and the more traditional applications. First we examine the new requirements that arise when considering control of VDT networks. We then provide a brief review of the capabilities and development status of each protocol. Finally, we examine possible evolution paths for the development of broadband signaling. Our primary objective is to describe a signaling evolution path that integrates the requirements of both business and residential broadband networks, thus allowing partial or total sharing of network resources in support of both types of applications     

Subscriber terminal units for video dial tone systems

The subscriber terminal unit (STU) receives and decodes video signals and provides interactive support for server-based multimedia applications delivered over a video dial tone network. The STU can be a maddening bottleneck or a gateway to the world. Done properly, it is the vital link that exploits the sophisticated capabilities of the broadband network. We describe the service model upon which applications are based and our current understanding of the application requirements. Different STU types, their capabilities, and expected availability are presented. A model of the functionality of a generic STU is presented, as well as several different development scenarios, including our approach to an actual implementation     

Applicability of ADSL to support video dial tone in the copper loop

An ADSL-3 architecture providing a downstream payload above 6 Mb/s has been proposed for operation over carrier serving area (CSA) loops. To determine whether such a system can be reliably deployed in the telephone company copper loop plant, a detailed model for loops and noise is presented. Results show that a sophisticated system, including echo cancellation and trellis coding, can reliably achieve CSA coverage, if T1 is not present in the plant. In the presence of assumed T1 worst case conditions, either the downstream bit rate must be reduced, or the operating range needs to be decreased to a serving area smaller than the CSA. A smaller “mid-CSA” serving area is proposed, and it is demonstrated that a 6.3-Mb/s ADSL (asymmetric digital subscriber line) system can operate in the presence of T1 in adjacent binder groups over this smaller serving area, which is roughly defined by 6 kilofeet of 26 gauge and 8 kilofeet of 24 gauge. A statistical study utilizing actual T1 line data suggests that if CSA coverage is attempted in the presence of T1 in adjacent binder groups, a success rate of greater than 92 percent can be expected     

MPEG-2 over ATM for video dial tone networks: issues and strategies

Market growth for PC multimedia and digital video owes largely to the rapid adoption of ISO compression standards by the industry. For video dial tone (VDT) services, the MPEG-2 set of standards have clearly emerged as the preferred coding method for VDT networks. For point-to-point switched video or multimedia connections, ATM has emerged as the technology of choice for switching and transport. This article describes how compressed digital video is transported over a VDT network, what some of the issues are, and how they are being addressed by the industry. It describes a generic VDT reference architecture, and the delivery method of video and multimedia information over such a network     

Local loop ISDN dial backup for frame relay

Corporations using FR services increasingly require access-line resiliency to their major carrier, and the carriers are, in return, offering a committment to an increased level of service. This article discusses a way of assuring that this service can be achieved. © 1998 John Wiley & Sons, Ltd.     

Direct-reading dial for noise temperature and noise resistance

An attenuator arrangement for a noise generator is described. The scheme permits direct reading of both noise resistance and noise temperature?the latter with a choice of source resistance.     

Dual-pilot tone calibration technique

Pilot-based calibration techniques are used to reduce the effects of multipath fading in mobile satellite receivers. One of the more recent of these techniques, namely the tone calibration technique (TCT), suggests transmitting double sideband modulation with the pilot tone located at the center of its spectrum where the amplitude and phase characteristics of the channel are most stable. To "make room" for the pilot in the presence of the Doppler shift, the equivalent low-pass data sidebands must be shaped so as to have zero response in the neighborhood of dc. Other techniques such as transparent tone-in-band (TTIB) similarly "notch out" a hole in the center of the data spectrum for location of the pilot. An alternate possibility which is at the same time much more bandwidth efficient than TCT is a dual-pilot tone calibration technique (DPTCT) that symmetrically locates a pair of pilots outside the data spectrum near the band edges of the channel. The operation and performance of DPTCT are analyzed, and its effectiveness is compared to that of the single tone TCT technique.     

基于图像处理的指针式仪表示数自动判读算法研究

本文为了获取指针式仪表的示数,研究了基于图像处理的示数自动判读算法。根据采集到的仪表示数图像,首先采用预处理增强、降采样及二值化、旋转投影指针的方法提取出指针所在的区域;然后选取备选区域并进行二值化处理,在备选区域中定位了针尖的位置;最后对刻度线进行标记和排序,完成了示数判读。结果表明,测量的214幅图像中,误差小于仪表最小分度值5%的读数占85.05%。本文网络版地址：http：//www. eepw.com.cn/article/274754.htm     

Enabling SIP-based sessions in ad hoc networks

The deployment of infrastructure-less ad hoc networks is suffering from the lack of applications in spite of active research over a decade. This problem can be solved to a certain extent by porting successful legacy Internet applications and protocols to the ad hoc network domain. Session Initiation Protocol (SIP) is designed to provide the signaling support for multimedia applications such as Internet telephony, Instant Messaging, Presence etc. SIP relies on the infrastructure of the Internet and an overlay of centralized SIP servers to enable the SIP endpoints discover each other and establish a session by exchanging SIP messages. However, such an infrastructure is unavailable in ad hoc networks. In this paper, we propose two approaches to solve this problem and enable SIP-based session setup in ad hoc networks (i) a loosely coupled approach, where the SIP endpoint discovery is decoupled from the routing procedure and (ii) a tightly coupled approach, which integrates the endpoint discovery with a fully distributed cluster based routing protocol that builds a virtual topology for efficient routing. Simulation experiments show that the tightly coupled approach performs better for (relatively) static multihop wireless networks than the loosely coupled approach in terms of the latency in SIP session setup. The loosely coupled approach, on the other hand, generally performs better in networks with random node mobility. The tightly coupled approach, however, has lower control overhead in both the cases. This work was partially done while the author was a graduate student in CReWMaN, University of Texas at Arlington. Dr. Nilanjan Banerjee is a Senior Research Engineer in the Networks Research group at Motorola India Research Labs. He is currently working on converged network systems. He received his Ph.D. and M.S. in computer science and engineering from University of Texas at Arlington. He received his B.E. degree in the same discipline from Jadavpur University, India. His research interests include telecom network architectures and protocols, identity management and network security, mobile and pervasive computing, measures for performance, modeling and simulation, and optimization in dynamic systems. Dr Arup Acharya is a Research Staff Member in the Internet Infrastructure and Computing Utilities group at IBM T.J. Watson Research Center and leads the Advanced Networking micropractice in On-Demand Innovation Services. His current work includes SIP-based services such as VoIP, Instant Messaging and Presence, and includes customer consulting engagements and providing subject matter expertise in corporate strategy teams. Presently, he is leading a IBM Research project on scalability and performance of SIP servers for large workloads. In addition, he also works on different topics in mobile/wireless networking such as mesh networks. He has published extensively in conferences/journals and has been awarded seven patents. Before joining IBM in 2000, he was with NEC C&C Research Laboratories, Princeton. He received a B.Tech degree in Computer Science from the Indian Institute of Technology, Kharagpur and a PhD in Computer Science from Rutgers University in 1995. Further information is available at Dr. Sajal K. Das is a Professor of Computer Science and Engineering and also the Founding Director of the Center for Research in Wireless Mobility and Networking (CReWMaN) at the University of Texas at Arlington (UTA). His current research interests include sensor networks, resource and mobility management in wireless networks, mobile and pervasive computing, wireless multimedia and QoS provisioning, wireless internet architectures and protocols, grid computing, applied graph theory and game theory. He has published over 400 research papers in these areas, holds four US patents in wireless internet and mobile networks. He received Best Paper Awards in IEEE PerCom’06, ACM MobiCom’99, ICOIN’02, ACM MSwiM’00 and ACM/IEEE PADS’97. He is also recipient of UTA’s Outstanding Faculty Research Award in Computer Science (2001 and 2003), College of Engineering Research Excellence Award (2003), the University Award for Distinguished record of Research (2005), and UTA Academy of Distinguished Scholars Award (2006). He serves as the Editor-in-Chief of Pervasive and Mobile Computing journal, and as Associate Editor of IEEE Transactions on Mobile Computing, ACM/Springer Wireless Networks, IEEE Transactions on Parallel and Distributed Systems. He has served as General or Program Chair and TPC member of numerous IEEE and ACM conferences. He is a member of IEEE TCCC and TCPP Executive Committees.     

Network level perspective in web sessions troubleshooting

In the last years, the quantity of data and the number of applications carried over web traffic have been continuously increasing and nowadays web browsing accounts for most of the Internet traffic. In such a scenario, a poor browsing experience can result very annoying to the end user, and the effective identification of the root cause of such bad performance is of primary interest to both the users and the network operators. In this paper, we present a unified framework, based on a novel lightweight open‐source publicly available probe and on an original statistical diagnosis algorithm, to correctly and effectively point out the segment of a web connection (eg, local client, backbone network, and DNS server) responsible for a poor web browsing experience. The extensive experimental evaluation carried out in the paper demonstrates the effectiveness of the proposed approach to diagnose poor quality of experience at a large scale.     

特种车辆仪表指针提取

提出了一种基于检测和验证相结合的指针综合提取方案,目的是实现特种车辆指针式仪表图像识别的自动识别和自动矫正。该方案优化利用了Hough变换法、极坐标中心投影以及边缘点集合的最小二乘法等算法,使得指针的自动提取无论针对何种品质的图像都非常准确。经验证,采用该方法进行特种车辆仪表指针提取,准确度达到99%以上。     

Protecting multicast sessions in WDM optical mesh networks

Recent advances in wavelength-division-multiplexing (WDM) technology are expected to facilitate bandwidth-intensive multicast applications. However, a single fiber (bundle) cut on such a network can disrupt the transmission of information to several destination nodes on a "light tree"-based multicast session. Thus, it is imperative to protect multicast sessions e.g., by reserving resources along backup trees. We show that, if a backup tree is directed-link-disjoint to its primary counterpart, then data loss can be prevented in the event of any single link failure. We provide mathematical formulations for efficient routing and wavelength assignment (RWA) of several multicast sessions (including their backup trees for dedicated protection) at a globally optimum cost. We present these formulations for networks equipped with two kinds of multicast-capable switch architectures: one using the opaque (O-E-O) approach and the other using transparent (all-optical) approach. We expand our formulations to accommodate sparse splitting constraints in a network, in which an optical splitter has limited splitting fanout and each node has a limited number of such splitters. We develop a profit-maximizing model that would enable a network operator to be judicious in selecting sessions and simultaneously routing the chosen ones optimally. We illustrate the solutions obtained from solving these optimization problem formulations for a representative-size network.     

Efficient allocation of elastic interfederation encrypted streaming sessions

Audio and video (A/V) collaboration platforms often use Internet cloud technologies to ensure elasticity. They generally operate on a best‐effort basis, without quality or delivery guarantees. However, such guarantees are a premise of business‐focused platforms, which often rely on static/dedicated infrastructure and hardware‐based components. This article presents results obtained in the final stage of the Elastic Media Distribution (EMD) project, which targets the migration of a business‐focused hardware‐based A/V collaboration tool towards a more elastic, reliable, and secure cloud‐based model. The use case under investigation is an educational scenario: teachers and students located at distributed sites collaborate under different data encryption policies. An existing model of collaboration streaming is extended to accommodate encryption‐enabled streaming components, and new resource allocation heuristics are proposed to deploy these components under stringent service level agreement (SLA) constraints. An extended version of our evaluation framework, based on the CloudSim simulator, manages encryption‐enabled components. A resource usage dataset was obtained by prototyping selected streaming components and evaluating their performance on the Virtual Wall large‐scale test bed. This dataset is fed into the extended simulation framework. Simulation results show longer than expected delays when loading streaming components, an issue that jeopardises the user experience that can be alleviated by the algorithms proposed in this article. Results show that the proposed algorithms enable policy‐based secured communications under bandwidth and virtual machine (VM) cost increases of 48% and 23%, respectively, if compared with a nonencrypted previous solution, and with set‐up times remaining under the required 2‐second deadline.     

Take off, plug in, dial up [aicraft personal communication]

The author describes how commercial airlines are now looking to deploy a slew of new in-flight technologies for business as well as entertainment. He explains how e-mail, Web surfing, cell phones and more will be coming to a plane near you     

"音质"和"音色"

“音质”和“音色”是两个不同的术语,在某些场合极易混用。如通常说一个扬声器的音质很好,但一般不说扬声器的音色很好;说一个音乐厅、歌剧院的音质设计得很好,并不说音乐厅和歌剧院的音色很好。而在另一些场合用“音色”这个术语时,经常是说“小提琴音色”、“大提琴音色”、“钢琴音色”等等,在电子琴中也用     

Admission control algorithms for multicast sessions with multiplestreams

Real-time multicast communication is considered, in which each destination makes an individual bid for delivery of a subset of real-time hierarchically encoded streams that are offered to the session by the source. The objective is to customize stream delivery to destinations, based on their requests and network constraints. Admission control procedures, which consider requests for multiple streams from multiple destinations and resolve contention when users' requests exceed available network resources, are presented. Specifications for four admission control procedures are presented and their properties analyzed. A performance measure for admission control is defined to be the total gain for the source that equals the sum of bids by the users taking delivery of the streams they ordered. Simulation results of the performance of the procedures are presented and compared to the performance of an optimal, but highly complex, admission control algorithm. In this paper the assumption is that the routing tree is already in place and the focus is on managing users' requests and deciding on which stream to deliver to each destination. This is done by signaling protocols that transfer messages containing users' requests and resource allocation commands between a source and the destinations. In addition, admission control protocols take users' requests and network resources as their input and issue as outputs the decisions on which request to admit (and allocate resources for) and which to reject     

一种基于视频指纹的视频检索方法研究

通过分块的色彩直方图对视频进行分割,提取每个视频子片段的关键帧,并提出了一种以SIFT特征为基础的语义词袋方法实现视频指纹的提取,该方法既提高了视频指纹提取的速度,又能得到具有较高独立性和鲁棒性的视频指纹。利用串匹配的方法实现视频指纹的匹配检测,并进一步对相似度进行加权处理,提高了匹配效率与匹配精度。