ITS无线接入网业务建模和调度方法设计

首先介绍了面向智能交通系统(Intelligent Transportation System,ITS)的无线接入网网络结构和主要业务,对车路协同业务和PTT语音业务进行了分析和建模。然后根据不同业务的特点及QoS需求提出具有针对性的调度方案:对于车路协同业务,采用适应车速的动态调度方法;对于PTT语音业务,采用基于状态转换的调度方法;对于视频和其他对时延不敏感的数据业务,采用缓存请求的调度方式。最后在OPNET平台上对调度方法进行仿真,针对各种ITS业务得出延时、丢包率等性能曲线,验证了算法的有效性,这也为ITS接入网的网络设计提供了参考。     

Direct push–pull or assisted push–pull? Toward optimal video content delivery using shared storage-based cloud CDN (SS-CCDN)

Video content delivery networks face many challenges such as scalability, quality of service and flexibility. Video suppliers address them through CDN. Cloud computing and Video content Delivery as a Service (VDaaS) plays a key role in improving the content delivery standard and makes the work of content providers, easier. By hosting video contents in the cloud, the content delivery costs are minimized and the overall content delivery performance enhanced by optimization of cloud CDN. Cost optimization of the cloud-based content delivery network requires a focus on delay or throughput, the overall performance and content delivery. The content placement and content access, the QoS and the QoE in CDN can be improved by enhancing the video content delivery performance. In this paper, a unique model for video content delivery, cloud-based is developed, titled as shared storage-based cloud CDN (SS-CCDN) to achieve the objective. This design optimizes through algorithms, the effective placement of video data and dynamic update of video data. For analysis, GA, PSO, and ACO algorithms are used. The proposed model uses direct and assisted push–pull content delivery schemes for cost-efficient content delivery. The low-cost VDaaS model reduces the storage cost, keeps the latency and the traffic cost. Experimental results validate that this model, with regard to storage, traffic, and latency generate higher performance with lower price and satisfy the QoS and QoE aspects in content delivery.

     

影响多媒体服务器性能的关键因素研究

在构建大规模视频服务系统时 ,基于层次型多服务器群的体系结构在吞吐率、可扩展性、经济性等方面都有其突出的优势 ,尤其适合于在因特网上的应用 .但是 ,要充分发挥和提高视频服务系统的性能 ,还要针对一些主要的瓶颈(如服务器磁盘 I/ O带宽与网络带宽 ) ,解决好一系列的问题 .本文分析了影响多媒体视频服务器性能的一些主要因素 ,如视频服务器的体系结构、服务器与客户端之间的数据传送方式、媒体数据在视频服务器存储子系统中的分布与放置方式、对磁盘访问请求的调度、单服务器中的缓存及多服务器间协同缓存的管理、接入控制策略、流调度策略等 ,这些因素对视频服务器的性能与吞吐率有着极大的影响 .本文还介绍了一些适用于大规模视频服务系统的性能优化技术 ,如广播、批处理等流调度策略 .在构建视频服务器系统时 ,只有综合考虑这些因素 ,才能真正提高服务器乃至整个视频服务系统的吞吐率 ,并较好地满足客户的 Qo S要求     

Application-aware scheduling for VoIP in Wireless Mesh Networks

Wireless Mesh Networks (WMNs) are seen as a means to provide last mile connections in Next Generation Networks (NGNs). Because of their auto-configuration capabilities and the low deployment cost WMNs are considered to be an efficient solution for the support of multiple voice, video and data services in NGNs. This paper looks at the optimal provision of resources in WMNs for Voice over IP (VoIP) traffic, which has strict performance requirements in terms of delay, jitter and packet loss. In WMNs, because of the challenges introduced by wireless multi-hop transmissions and limited resources, providing performance quality for VoIP comparable to the voice quality in the traditional circuit-switched networks is a major challenge.This paper analyses different scheduling mechanisms for TDMA-based access control in mesh networks as specified in the IEEE 802.16-2004 WiMAX standard. The performance of the VoIP applications when different scheduling mechanisms are deployed is analysed on a variety of topologies using ns-2 simulation and mathematical analysis. The paper concludes that on-demand scheduling of VoIP traffic – typically deployed in 802.11-based WMNs – is not able to provide the required VoIP quality in realistic mesh WiMAX network scenarios and is therefore not optimal from a network operator’s point of view. Instead, it is shown, that continuous scheduling is much better suited to serve VoIP traffic. The paper then proposes a new VoIP-aware resource coordination scheme and shows, through simulation, that the new scheme is scalable and provides good quality for VoIP service in a wide range of network scenarios. The results shown in the paper prove that the new scheme is resilient to increasing hop count, increasing number of simultaneous VoIP sessions and the background traffic load in the network. Compared to other resource coordination schemes the VoIP-aware scheduler significantly increases the number of supported calls.     

无线视频传输服务质量研究

在下一代无线通信网络中,以视频传输为主的流媒体业务将成为主要业务之一。着眼于无线视频流服务质量的研究,首先简要介绍了以网络为中心的服务质量控制策略,然后重点分析以终端为中心的速率控制机制,并介绍了目前常用的分组调度算法。     

A dynamic overlay approach for mobility maintenance in personal communication networks

The location predictions of the nodes are the key issues in mobile environment. The applications like video sharing, voice over IP and other online applications offer better results with stable nodes. In mobile scenario, connectivity and quality of service are critical issues. To overcome these issues dynamic overlay approaches are proposed in this work namely, 1.Mobile Peers-Unstructured, 2.Structured Peer, 3.MANET Structure and 4.Mesh Backbone. These structures are obtained from the network, during the requirement of connectivity or quality of service. The performance of the proposed methods are analyzed in stable and mobile conditions by analyzing power consumption, hop stretch, packet delivery ratio, network life time and other QoS parameters.     

ViMediaNet: an emulation system for interactive multimedia based telepresence services

The telepresence service (TPS) leverages advanced communication, virtual reality, and cooperative web technologies to provide remote engaged users an immersive experience of being fully present, which are widely used in teleconferencing, remote surgery, and hazardous environment exploration. The common feature of TPS-based applications lies in that the users’ experiences heavily depend on the quality of interactive multimedia services, that is, most deliveries of multimedia contents are mission-critical, as well as time-critical, to ensure the control and operation accuracy for telerobots. Therefore, the delivery performance of multimedia contents over heterogeneous networks becomes the primary concern in the TPS system design, especially for those delay sensitive applications, like remote surgery and environment exploration. In this paper, we incorporate video processing tools with the network simulator to design and implement a joint emulation system, named as the virtual media network (ViMediaNet), through which, the TPS designer can readily deploy the TPS system over a heterogeneous network virtualized by the network emulator, investigate the critical network performance metrics in the multimedia content delivery, such as transmission delay and packet jitter, and observe control results in real time. To verify the capability of ViMediaNet in emulating the real-setting TPS system, we take a mobile device based vision navigation system as an example to evaluate performance impacts of the rate adaptation technology in the telerobot control. The experimental results reveal that the adaptive video streaming is a promising solution to significantly enhance the control accuracy of telerobots in TPS systems.     

Modeling and evaluating IPTV applications in WiMAX networks

Internet Protocol-based Television (IPTV) is a digital television service which delivers television content via an IP network. The rapid growth of wireless network technology in recent years has changed, the way people access the Internet. Adding mobility to IPTV can create a truly compelling ubiquitous service which spans different network domains and varied IP-enabled terminals and devices, such as set-top boxes, PCs and cell phones. However, extending IPTV service to wireless networks requires overcoming bandwidth bottlenecks and high packet loss rates. Following the IEEE 802.16 standard, worldwide interoperability for microwave access (WiMAX) features high data rates and large service coverage, offering a wireless broadband solution for IPTV services. While previous research has focused on creating a broadband IPTV service few studies have practically evaluated IPTV applications in a wireless broadband network environment. In this paper, we model and evaluate a common constant bit rate (CBR)¹ based IPTV application and an IPTV live streaming (PPStreaming)² application while retrieving IPTV content via a WiMAX network. We also use the NS2 simulation tool to evaluate the performance of these two IPTV applications. The evaluation metrics include latency, packet loss, data rate and throughput statistics when the two IPTV applications are run in the WiMAX network. ¹The simplest IPTV solution is to convey video content by CBR. IPTV operators and content delivery networks relay CBR streaming content to control the demand for network capacity. Broadcasters prefer CBR video as it conserves bandwidth resources, but CBR delivery can degrade video quality result in higher overall demand on network capacity. ²PPStreaming (also referred to as P2P streaming Internet TV) is a network for live media streaming. In principle it’s similar to BitTorrent (BT) in that it provides stable and smooth broadcast of TV programs to broadband users. Unlike traditional streaming media, PPStreaming adopts P2P-streaming technology and supports full-scale visits with tens of thousands of simultaneous users. Its client software can be used in the browser or as a standalone executable.     

Performance analysis of a pull-based parallel video server

In conventional video-on-demand systems, video data are stored in a video server for delivery to multiple receivers over a communications network. The video server's hardware limits the maximum storage capacity as well as the maximum number of video sessions that can concurrently be delivered. Clearly, these limits will eventually be exceeded by the growing need for better video quality and larger user population. This paper studies a parallel video server architecture that exploits server parallelism to achieve incremental scalability. First, unlike data partition and replication, the architecture employs data striping at the server level to achieve fine-grain load balancing across multiple servers. Second, a client-pull service model is employed to eliminate the need for interserver synchronization. Third, an admission-scheduling algorithm is proposed to further control the instantaneous load at each server so that linear scalability can be achieved. This paper analyzes the performance of the architecture by deriving bounds for server service delay, client buffer requirement, prefetch delay, and scheduling delay. These performance metrics and design tradeoffs are further evaluated using numerical examples. Our results show that the proposed parallel video server architecture can be linearly scaled up to more concurrent users simply by adding more servers and redistributing the video data among the servers     

An applied research study for the provision of a QoS-oriented environment for voice and video services over satellite networks

The aim of this paper is to present a network solution to provide Quality of Service (QoS) guaranteed voice and video services over a satellite test-bed based on the TCP/IP protocol suite, oriented to distance learning. The scope of the solutions proposed is a small private satellite network where specific services (e.g. tele-education) are offered to the users.The paper describes the main operating steps followed to perform the tests. The target service requirements are identified and the configuration of devices and protocols are described in detail. The Integrated Services approach is utilized along with three different scheduling mechanisms at IP layer. A proper routing protocol to support multicast is used.The work is a part of a project called ‘Integration of Multimedia Services over Heterogeneous Satellite Networks’, which has been developed by the Italian National Consortium of Telecommunications (CNIT) and funded by the Italian Space Agency (ASI). The test-bed is composed of two remote sites, connected through a satellite link at 2 Mbits/s and by one site, where no satellite device is available, connected by using ISDN.The results reported allow to test the application of bandwidth reservation strategies originally studied for a terrestrial environment and to measure the consequent performance improvement. The use of both subjective metrics as mean opinion score (MOS) and objective metrics, as audio/video packet loss and average bit rate, to perform the investigation, allow mapping the measure of P-QoS (Perceived-Quality of Service) with quantities uninfluenced by personal opinions. The configurations used represent the result of an applied research project and provide an operative solution, inclusive of technical details, to guarantee high quality of service for audio/video applications.     

Fine-Granularity Transmission Distortion Modeling for Video Packet Scheduling Over Mesh Networks

Packet scheduling is a critical component in multi-session video streaming over mesh networks. Different video packets have different levels of contribution to the overall video presentation quality at the receiver side. In this work, we develop a fine-granularity transmission distortion model for the encoder to predict the quality degradation of decoded videos caused by lost video packets. Based on this packet-level transmission distortion model, we propose a content-and-deadline-aware scheduling (CDAS) scheme for multi-session video streaming over multi-hop mesh networks, where content priority, queuing delays, and dynamic network transmission conditions are jointly considered for each video packet. Our extensive experimental results demonstrate that the proposed transmission distortion model and the CDAS scheme significantly improve the performance of multi-session video streaming over mesh networks.     

Scheduled video delivery-a scalable on-demand video delivery scheme

Continuous media, such as digital movies, video clips, and music, are becoming an increasingly common way to convey information, entertain and educate people. However, limited system and network resources have delayed the widespread usage of continuous media. Most existing on-demand services are not scalable for a large content repository. In this paper, we propose a scalable and inexpensive video delivery scheme, named Scheduled Video Delivery (SVD). In the SVD scheme, users submit requests with a specified start time. Incentives are provided so that users will specify the start times that reflect their real needs. The SVD system combines requests to form the multicasting groups and schedules these groups to meet the deadline. SVD scheduling has a different objective from many existing scheduling schemes. Its focus has been shifted from minimizing the waiting time toward meeting deadlines and at the same time combining requests to form multicasting groups. SVD compliments most existing video delivery schemes as it can be combined with them. It requires much less resources than other schemes. Simulation study for the SVD performance and the comparison to other schemes are presented.     

运动感知基于缓存的自适应视频流传输

目的 基于缓存的自适应视频流传输策略无需估测实时带宽,直接通过缓存变化量与码率的映射函数选取符合当前网络状况的最佳质量码流传输。传统基于缓存的自适应视频传输不考虑内容特征,在码率选择上为不同运动级别视频内容均使用相同的码率映射函数,在不稳定的无线网络环境中高运动强度内容的码率急剧降低会严重伤害用户体验质量（QoE）,提出运动感知基于缓存的自适应视频流传输（MA-BBA）算法。方法 MA-BBA算法根据片段运动级别确定码率映射函数,对运动强度高的内容快速切换到较高码率,而对于运动强度较低的内容则使用较为保守的码率,从而使得缓存资源能够位于安全边界之上且较多分配给高级别运动内容,提高不同运动强度内容的平均质量,使整体QoE得到优化。结果 在公开的无线网络带宽数据集上实现本文MA-BBA算法,基于吞吐量的自适应传输算法（TBA）和基于缓存的自适应传输算法（BBA）。MA-BBA在高运动强度内容的平均质量上比TBA和BBA分别提高1.7%和1.2%,且质量波动区间更小。MA-BBA在平均缓存利用率上达到72%,大大高于TBA的45.9%和BBA的45.4%。结论 MA-BBA算法与现有的码率自适应算法TBA和BBA相比,大大提高了缓存资源利用率,提高了对资源要求最苛刻的高级别运动内容的传输质量,减小码率切换幅度频率,优化了视频服务的整体QoE。     

Seamless handover between unicast and multicast multimedia streams

With the deployment of heterogeneous networks, mobile users are expecting ubiquitous connectivity when using applications. For bandwidth-intensive applications such as Interact Protocol Television （IPTV）, multimedia contents are typical- ly transmitted using a multicast delivery method due to its bandwidth efficiency. However, not all networks support multicasting. Multicasting alone could lead to service disruption when the users move from a multicast-capable network to a non-multicast network. In this paper, we propose a handover scheme called application layer seamless switching （ALSS） to provide smooth real-time multimedia delivery across unicast and multicast networks. ALSS adopts a soft handover to achieve seamless playback during the handover period. A real-time streaming testbed is implemented to investigate the overall handover performance, espe- cially the overlapping period where both network interfaces are receiving audio and video packets. Both the quality of service （QoS） and objective-mapped quality of experience （QoE） metrics are measured. Experimental results show that the overlapping period takes a minimum of 56 and 4 ms for multicast-to-unicast （M2U） and unicast-to-multicast （U2M） handover, respectively. The measured peak signal-to-noise ratio （PSNR） confirms that the frame-by-frame quality of the streamed video during the handover is at least 33 dB, which is categorized as good based on ITU-T recommendations. The estimated mean opinion score （MOS） in terms of video playback smoothness is also at a satisfactory level.     

Zebroid: using IPTV data to support STB-assisted VoD content delivery

IPTV, unlike Internet TV, delivers digital TV and multimedia services over IP-based networks with the required level of quality of service (QoS) and quality of experience (QoE). Linear programming channels in IPTV are delivered through multicast, which is highly scalable with the number of subscribers. Video-on-demand (VoD) content, on the other hand, is typically delivered using unicast, which places a heavy load on the VoD servers and all the network components leading to the end-user set-top boxes (STBs) as the demand increases. With the rapid growth of IPTV subscribers and the shift in video viewing habits, the need to efficiently disseminate large volumes of VoD content has prompted IPTV service providers to consider the use of STBs to assist in video content delivery. This paper describes our current research work on Zebroid, a potential VoD solution for fiber-to-the-node (FTTN) networks, which uses IPTV data on a recurring basis to determine how to select, stripe, and preposition popular content in selected STBs during idle hours. A STB requesting VoD content during the peak hours can then receive necessary stripes from participating STBs in the neighborhood. Recent VoD request access patterns, STB availability data, and capacity data on network components are taken into consideration in determining the parameters used in the striping algorithm of Zebroid. We show both by simulation and emulation on a realistic IPTV testbed that the VoD server load can be reduced by more than 70% during peak hours by allocating only 8 GB of storage on each STB. The savings achieved through Zebroid would also allow IPTV service providers to add more linear programming channels without expensive infrastructure upgrades.     

不同队列管理机制对多媒体传输品质的影响

Evalvid是一套用于评估网络仿真环境下多媒体视频传输质量的工具,应用其在网络仿真软件NS中的扩展实现,仿真分析了采用不同的网络队列管理机制对多媒体传输质量的影响.分析的队列管理机制包括丢尾杌制、主动队列管理机制、差异式服务机制;并利用丢包率、PSNR对多媒体传输质量进行了分析.结果显示主动队列管理机制和差异式服务机制可以很好地保证多媒体传输的质量.     

Advances in voice quality measurement in modern telecommunications

Quality of service (QoS) is a measure of a communication network performance that reflects the transmission quality and service availability as perceived by the users. In the context of telecommunications, voice communication quality is the most visible and important aspects to QoS, and the ability to monitor and design for this quality should be a top priority. Voice quality refers to the clearness of a speaker's voice as perceived by a listener. Its measurement offers a means of adding the human end-user's perspective to traditional ways of performing network management evaluation of voice telephony services. Traditionally, measurement of users' perception of voice quality has been performed by expensive and time-consuming subjective listening tests. Over the last decade, numerous attempts have been made to supplement subjective tests with objective measurements based on algorithms that can be computerised and automated. This paper examines some of the technicalities associated with voice quality measurement, and presents a review of current subjective and objective voice quality measurement methods and standards as applied to telecommunication systems and devices, with particular focus on recent and internationally standardised methods.     

中国高性能宽带信息网环境下视频点播业务模型

针对因特网无法提供电信级服务质量保证的问题,提出了中国高性能宽带信息网(3Tnet)环境下一种大规模视频点播(Video-on-Demand,VoD)业务模型.该模型适应实时播放及高清晰回放需求,在骨干网采用内容分发网络,在接入网采用P2P策略.讨论了该模型的可行性,给出了性能评估.仿真结果证明该模型支持3Tnet上大规模VoD业务.     

ORN: A content-based approach to improving supplier discovery in P2P VOD networks

There are two major building blocks in operating a peer-to-peer (P2P) video-on-demand (VOD) network: supplier discovery and content delivery. Supplier discovery concerns the discovery of peer nodes in the network that can provide the streaming data blocks needed for playing by a local node. The more suppliers one can discover, the higher the chance of locating quality suppliers for delivering contents smoothly to ensure uninterrupted playback. The key to supplier discovery is to establish and track the supply-demand relationship among the peers. For P2P VOD, the supply-demand relationship is determined by the buffer contents of the peers. Unfortunately, the buffer contents change rapidly as peers play the video, especially under VCR operations. The challenge is to track all the dynamic relationships in an efficient way. In this paper, we propose an Overlapping Relation Network (ORN). The idea is to track the dynamic supply-demand relationship by tracking the overlapping of peers’ buffer contents. As long as peers play the video at the same rate, the overlapping relationship is stable and can be used for low-cost supplier discovery. Extensive analyses and simulation experiments show that in most cases the ORN can discover more than 96% of the suppliers in the network, resulting in a streaming continuity that is superior to that of other approaches.     

QinQ在城域网中的应用浅析

随着互联网业务的不断发展,IP网络上新应用、新技术的不断涌现,对网络的质量服务提出了新的要求。IP城城网作为综合业务接入网络,将从传统的数据业务接入向语音、视频和数据等多业务综合接入平台转化。根据商业运营发展的需求,本文介绍了灵活QinQ技术概念、特性及在城域网环境下的应用。