基于端系统的视频流组播速率控制方法

针对目前Internet未对多媒体应用提供QoS保障的问题,分析了视频流组播的难点,提出了一种基于缓冲区管理的网络自适应组播发送速率控制方法.该机制可以合理控制服务器的发送速率,既能自适应网络状况的变化,又满足流媒体实时播放的需求.实验结果表明,该机制通过控制发送缓冲区占有率,降低了分组丢包率,提高了终端的接收质量,具有良好的实用价值.     

基于客户端缓冲区预警界限的流媒体传输速率控制方法

介绍了一种新的基于速率的网络拥塞控制方法。该控制方法通过监测网络状态变化和客户端缓冲区状态,动态调整服务器端的发送速率,能够很好提高服务质量。通过研究表明,该方法在流媒体网络传输中能够取得很好的效果。     

流媒体网络传输中基于速率的控制方法

介绍了一种新的基于速率的拥塞控制方法，重点讨论了速率控制方法在流媒体网络传输中的具体应用。该方法通过监测网络状态变化和客户端缓冲区状态，动态调整服务器端的发送速率，以提高服务质量。研究表明，相对于传统的基于窗口型拥塞控制方法，速率控制方法在流媒体网络传输中可取得更好的效果。     

一种模糊控制的组播速率调节机制

目前多数组播拥塞控制机制采用模拟TCP窗口机制传输流媒体业务,尽管保证了TCP友好性,但是速率不够平滑,不能很好地满足流媒体组播业务服务质量的要求。针对这一问题,提出了一种模糊控制的组播速率调节算法（FC-MRAA）。该算法基于模糊控制理论设计了两个模糊控制器,一个根据接收端的反馈信息计算速率增量,保证TCP友好性;另一个根据路由器缓冲区占有率计算控制增益,平滑发送速率。仿真结果表明,该算法具有良好的速率平滑性和TCP友好性。     

基于RTCP的移动流媒体研究

在移动网络发生拥塞时,由于传统的TCP友好速率控制机制(TFRC)存在丢包率偏差问题,导致移动流媒体服务器不能获取正确的嗣络状态信息,从而过度调节发送速率.为了解决问题,提出一种基于RTCP的改进机制.基于移动网络的特性,机制结合时间分布均匀化的RTCP发送机制缓解RTCP初始化洪泛,通过分析RTCP报告,在流媒体服务器端正确区分出网络拥塞丢包和移动网络随机误码丢包,以获得正确的丢包率,最终实现数据发送速率的自适应调节.利用NS2网络仿真软件进行仿真,结果显示该机制比TFRC机制更具备适应网络波动的能力,并能维持较高的网络利用率.移动流媒体的服务质量得到提升.     

基于模糊逻辑的单速率组播拥塞控制机制

分析了现有的单速率组播拥塞控制的优缺点,提出了一种基于模糊逻辑的单速率组播拥塞控制机制FSRMCC,在FSRMCC中接收端利用指数平滑预测模型预测期望速率,运用模糊判决器控制反馈信息的发送,减少反馈包的数量,发送端根据反馈信息通过模糊控制器及时调整发送速率,减少速率抖动;仿真表明,FSRMCC具有良好的TCP友好性、速率平滑性、响应性以及可扩展性,适用于流媒体组播业务的传输.     

流媒体系统中基于速率的拥塞控制方法研究

在比较和研究各类基于速率的拥塞控制机制的基础上．构建出一种新的流媒体系统中的速率型拥塞控制方法。该方法通过监测网络状态变化和客户端缓冲区状态．动态调整服务器端的发送速率,减少包丢失率和缩短延迟时间,提高系统服务质量。此方法表现出很好的灵活性和稳定性,计算复杂度很低,不影响传输效率。和传统的拥塞控制机制相比,此方法在控制的实时性、网络状态震荡抑制、多媒体通信质量等方面均有一定的改善和提高。     

基于种群生态的组播拥塞控制机制

针对现有TCP类组播拥塞控制机制不具有速率平滑性、往返时间(RTT)公平性以及在高速环境中传输效率低的问题,提出一种基于种群生态理论的自适应高速组播拥塞控制机制。该机制在每个接收端实现瓶颈链路带宽和背景流速率的测量,并将这两个测量值用于种群生态模型中以计算期望服务速率,然后使用一种简单的反馈抑制机制选取期望服务速率最小的接收端作为代表,该代表将其期望服务速率反馈给源端控制发送速率。仿真结果表明新机制发送速率平滑,具有RTT公平性,在低速网络和高速网络中都能与单播流公平共享带宽资源。     

松弛算法在TCP友好速率控制中的应用

由于TCP友好速率控制（TFRC）机制在实时多媒体应用中，TFRC流的发送速率波动性明显，不利于实时多媒体流的传输。采用松弛算法对TFRC流的发送速率进行自适应约束，使TFRC流发送速率在与TCP流竞争中变得更加平缓、收敛。实验结果表明，松弛算法能够改善TFRC的性能，提高实时流媒体的传输质量。     

组播中层次化视频的传输

1.简介由于Internet网络本身的限制,使多点通信成为一个相对较复杂的问题:因为各个组播对象的条件不同,使发送者无法通过发送固定带宽数据的方法来满足所有接收者的要求。为了达到尽可能地利用已有网络资源的目的,就需要对视频组播的速率进行控制。针对这个问题,有两种解决方法:由发送者进行控制,     

Range Multicast for Video on Demand

We explore a communication paradigm for video on demand, called Range Multicast. This schemeis a shift from the conventional thinking about multicast where every receiver must obtain the same data packet at any time. A range multicast allows new members to join at their specified time and still receive the entire video stream without consuming additional server bandwidth. Clients enjoy better service latency since they can join an existing multicast instead of waiting for the next available server stream. We also present techniques to support video-cassette-recorder-like interactivity in this environment. Unlike existing methods which require clients to cache data in a private buffer, the Range Multicast solution utilizes the shared network storage to make more efficient and cost-effective use of the caching space. Furthermore, since a range multicast can accommodate clients with different play points in the video, a client has a better chance to join an on-going multicast for normal playback after finishing a VCR operation. This strategy avoids the need for a new server stream, and thus further alleviates the server load. Our simulation results confirm the aforementioned benefits.     

Internet多媒体分层组播JSCC差错控制率失真优化

针对Internet多媒体群组通信中同时存在的带宽异构性和包丢失率异构性，文中将分层组播和接收者驱动的思想扩展到FEC差错控制中，提出一种分层FEC组播差错控制方法LM-FEC．LM-FEC通过不同的组播组发送信源编码层和各信源层的FEC校验数据，为接收者根据信道带宽和数据包丢失率实施差错控制提供更加灵活的选择．文中用FH-MDP模型描述接收者行为，通过JSCC率失真优化确定编码层内和编码层间的速率分配，JSCC率失真优化采用变量替换和动态规划算法求解．实验表明，该文提出的差错控制方法能够有效改善重建多媒体信号的回放质量．     

Buffer-assisted on-demand multicast for VOD applications

Multicast communications is widely used by streaming video applications to reduce both server load and network bandwidth. However, receivers in a multicast group must access the multicast stream simultaneously, and this restriction on synchronous access diminishes the benefit of multicast because users in a video-on-demand service usually issue requests asynchronously, i.e., at anytime. In this paper, we not only formulate this streaming problem but also propose a new multicast infrastructure, called buffer-assisted on-demand multicast, to allow receivers accessing a multicast stream asynchronously. A timing control mechanism is integrated on intermediate routing nodes (e.g., routers, proxies, or peer nodes in a peer-to-peer network) to branch time-variant multicast sub-streams to corresponding receivers. Besides, an optimal routing path and the corresponding buffer allocations for each request must be carefully determined to maximize the throughput of the multicast stream. We prove that the time complexity to solve this routing problem over general graph networks is NP-complete, and then propose a routing algorithm for overlay networks to minimize server load. Simulation results demonstrate that buffer-assisted on-demand multicast outperforms many popular streaming methods.     

Minimizing bandwidth requirements for on-demand data delivery

Two recent techniques for multicast or broadcast delivery of streaming media can provide immediate service to each client request, yet achieve considerable client stream sharing which leads to significant server and network bandwidth savings. The paper considers: 1) how well these recently proposed techniques perform relative to each other and 2) whether there are new practical delivery techniques that can achieve better bandwidth savings than the previous techniques over a wide range of client request rates. The principal results are as follows: First, the recent partitioned dynamic skyscraper technique is adapted to provide immediate service to each client request more simply and directly than the original dynamic skyscraper method. Second, at moderate to high client request rates, the dynamic skyscraper method has required server bandwidth that is significantly lower than the recent optimized stream tapping/patching/controlled multicast technique. Third, the minimum required server bandwidth for any delivery technique that provides immediate real-time delivery to clients increases logarithmically (with constant factor equal to one) as a function of the client request arrival rate. Furthermore, it is (theoretically) possible to achieve very close to the minimum required server bandwidth if client receive bandwidth is equal to two times the data streaming rate and client storage capacity is sufficient for buffering data from shared streams. Finally, we propose a new practical delivery technique, called hierarchical multicast stream merging (HMSM), which has a required server bandwidth that is lower than the partitioned dynamic skyscraper and is reasonably close to the minimum achievable required server bandwidth over a wide range of client request rates     

P2P transfer of partial stream in multimedia multicast

In this paper, we propose a new multicast scheme that is based on the client-initiated-with-prefetching (CIWP) and peer-to-peer (P2P) transfer of a partial multimedia stream. In the CIWP scheme, when a new client joins an ongoing multicast channel, the server has to create an extra unicast channel to retransmit the partial stream that has already been transmitted. However, the unicast channel consumes some of the I/O bandwidth of the server, as well as some of the network resources between the server and the client's Internet Service Provider (ISP). To solve this problem, we propose the use of the P2P transfer algorithm to deliver the partial stream from a client that has already joined the ongoing multicast session to the newcomer. This P2P transfer between clients is limited to clients belonging to the same ISP. To further improve the performance, a threshold is used to control the P2P transfer. We performed analytical studies to show that the proposed multicast scheme can reduce the consumption of the network resources of the server, by utilizing the client's disk space. We also performed various simulation studies to demonstrate the performance improvement in terms of the use of the server's bandwidth and the waiting time for the clients’ requests.     

流媒体传输中的QoS研究及其实现

文中提出的对视频会议的服务质量控制策略是为解决以下问题:视频会议对涉及到的实时数据流传输对网络带宽、时延和丢包率有较高要求,但是,目前得到广泛应用的IP网络是一种尽力而为的网络,它并没有对实时多媒体提供任何服务质量保证。该策略从两个方面对服务质量加以控制:在发送端控制发送流量;在数据再现端通过缓冲机制控制媒体同步。     

Reducing bandwidth requirement for delivering video over wide area networks with proxy server

Due to the high bandwidth requirement and rate variability of compressed video, delivering video across wide area networks (WANs) is a challenging issue. Proxy servers have been used to reduce network congestion and improve client access time on the Internet by caching passing data. We investigate ways to store or stage partial video in proxy servers to reduce the network bandwidth requirement over WAN. A client needs to access a portion of the video from a proxy server over a local area network (LAN) and the rest from a central server across a WAN. Therefore, client buffer requirement and video synchronization are to be considered. We study the tradeoffs between client buffer, storage requirement on the proxy server, and bandwidth requirement over WAN. Given a video delivery rate for the WAN, we propose several frame staging selection algorithms to determine the video frames to be stored in the proxy server. A scheme called chunk algorithm, which partitions a video into different segments (chunks of frames) with alternating chunks stored in the proxy server, is shown to offer the best tradeoff. We also investigate an efficient way to utilize client buffer when the combination of video streams from WAN and LAN is considered.     

基于双缓冲区的发送速率控制方法

分析发送端/接收端发送速率控制的相关研究,提出一种基于双缓冲区的发送速率控制方法。该方法能提高流媒体服务器的并发流数目,将媒体数据的发送速率控制在一定的范围内,保证用户所需的带宽资源不会出现剧烈震荡,从而使客户端能得到平稳的数据流,获得较好的视频质量。     

异构环境下层次编码多视频源多共享信道分层组播

视频组播是许多当前和将来网络服务的重要组成部分,如视频会议,远程学习、远地展示及视频点播,随着网络传送基础设施的改善和端系统处理能力的增强,组播视频应用日益变得可行,组播视频传输中存在的主要问题是网络送资源的异构性和动态性,其使得视频流的多个接收方都达到可接受的流量特性变得异常困难,目前该问题的一个有效解决方式就是利用自适应的分层视频传输机制,在该机制中,各源产生层次媒体流,并在多个网络信道中传输。对视频会议类的多点到多点视频组播应用,信道往往被所有潜在的发送方共享,任何发送方都可在任何一个共享信道中发送其视频层次。在该多点到多点、共享信道、分层视频组播模型下,一个关键问题就是如何动态确定各视频源层次到各共享组播信道的映射,映射策略直接影响到会话整体视频接收质量和网络带宽利用率。典型的方式是顺序映射,该映射方式同等对待各发送方,但利用该方式,随源数目的增加,在各共享网络信度上会出现带宽可伸缩性问题,而且顺序映射方式无法适应网络传送资源和会话状态的动态变化。为此,该文设计了一种基于接收方反馈信息的自适应的层次映射算法,接收方周期性地将其当前感兴趣的发送方及接收速率的信息反馈给某控制节点,而控制节点就利用当前反馈信息动态地调整映射策略。经证实,该算法始终能比顺序层次映射算法获得更高的整体视频接收质量,并具有高的带宽利用率和很小的复杂度。