公开(公告)号：US20060007947A1

公开(公告)日：2006-01-12

申请号：US10887406

申请日：2004-07-07

申请人： Jin Li ,  Philip Chou ,  Cha Zhang

发明人： Jin Li ,  Philip Chou ,  Cha Zhang

IPC分类号： H04J3/26

CPC分类号： H04L12/1877 ,  H04L12/1854 ,  H04L29/06027 ,  H04L65/607 ,  H04L67/06 ,  H04L67/104 ,  H04L67/1076 ,  H04L67/108 ,  H04L67/34 ,  H04L69/16 ,  H04L69/163 ,  H04L69/165

摘要： A content distribution method and system for distributing content over a peer-to-peer network such that the full potential throughput of the network is achieved. The content distribution method divides the content to be distributed into many small blocks. Each of the content blocks then is assigned to a node, which can be a content-requesting node, a non-content-requesting node or a source node. Content is assigned based on a capacity of the node, where nodes having a larger capacity are assigned a greater number of content blocks and nodes having a smaller capacity are assigned a fewer content blocks. The capacity generally is defined as the upload bandwidth of the node. Redistribution queues are employed to control the throughput of the distribution. This bandwidth control strategy ensures that upload bandwidths of the peer and source nodes are fully utilized even with network anomalies such as packet losses and delivery jitters.

公开(公告)号：US20070206610A1

公开(公告)日：2007-09-06

申请号：US11275177

申请日：2005-12-16

申请人： Dan Teodosiu ,  Philip Chou ,  Alexandra Heron ,  Cheng Huang ,  Tom Kleinpeter ,  Jin Li ,  Eyal Schwartz ,  John Spivey ,  David Steere

发明人： Dan Teodosiu ,  Philip Chou ,  Alexandra Heron ,  Cheng Huang ,  Tom Kleinpeter ,  Jin Li ,  Eyal Schwartz ,  John Spivey ,  David Steere

IPC分类号： H04L12/56 ,  H04J3/06

CPC分类号： H04L67/325 ,  H04L67/06 ,  H04L67/104 ,  H04L67/1044 ,  H04L67/1095 ,  H04W76/14

摘要： Various embodiments can utilize a distributed solution for scheduling connections between peers in a file sharing environment. In accordance with at least some embodiments, individual nodes in a peer-to-peer system include scheduling software that enables the nodes to make scheduling decisions with regard to how, when and where connections are made with other peers. Scheduling decisions can be based on a number of different parameters. In at least some embodiments, a synchronization negotiation manager is embodied with logic that drives the negotiation and scheduling process. This logic is represented by an innovative state machine that is designed to implement the negotiation and synchronization process.

摘要翻译： 各种实施例可以利用分布式解决方案来调度文件共享环境中的对等体之间的连接。 根据至少一些实施例，对等系统中的单个节点包括调度软件，其使节点能够就如何，何时何地与其他对等体进行连接做出调度决定。 调度决定可以基于许多不同的参数。 在至少一些实施例中，同步协商管理器具有驱动协商和调度过程的逻辑。 该逻辑由创新的状态机表示，旨在实现协商和同步过程。

公开(公告)号：US07596488B2

公开(公告)日：2009-09-29

申请号：US10663390

申请日：2003-09-15

申请人： Dinei Florencio ,  Philip Chou ,  Li-Wei He

发明人： Dinei Florencio ,  Philip Chou ,  Li-Wei He

IPC分类号： G10L19/04 ,  G10L21/04 ,  G10L11/06

CPC分类号： H04N21/4392 ,  G10L19/005 ,  H04N21/4394 ,  H04N21/4396 ,  H04N21/4398

摘要： An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.

摘要翻译： “自适应音频播放控制器”通过将音频信号的接收分组解码并读取到信号缓冲器来进行操作。 然后根据播放器设备的需要从信号缓冲器中播放经解码的音频信号的样本。 抖动控制和分组丢失隐藏是通过实时连续分析缓冲区内容来实现的，并且确定是否从缓冲器内容中提供未修改的重放，是否压缩缓冲区内容，扩展缓冲区内容，还是提供丢包隐藏 延迟或丢失的数据包作为缓冲区内容的函数。 此外，自适应音频重放控制器还确定在哪里拉伸或压缩信号缓冲器中的特定帧或信号段，以及拉伸或压缩这些段以便优化感知的播放质量。

公开(公告)号：US20050185578A1

公开(公告)日：2005-08-25

申请号：US10787612

申请日：2004-02-25

申请人： Venkata Padmanabhan ,  Jiahe Wang ,  Philip Chou

发明人： Venkata Padmanabhan ,  Jiahe Wang ,  Philip Chou

IPC分类号： G01R31/08 ,  H04L12/18 ,  H04L12/56

CPC分类号： H04L12/185 ,  H04L41/5009 ,  H04L41/5035 ,  H04L43/0829 ,  H04L47/10 ,  H04L47/11 ,  H04L47/12 ,  H04L67/104 ,  H04L67/1089

摘要： A technique is disclosed that can efficiently control congestion, while supporting heterogeneity for streaming data among multiple computers in a network. A plurality of nodes is divided into a plurality of distribution trees within a computer network, wherein the data is divided into a plurality of prioritized layers. When a node experiences packet loss, the location of the congestion is inferred. If the congestion is at or near the outgoing link, outgoing traffic is shed to alleviate the congestion by shedding child node(s) receiving descriptions in the least important layer of data that the child node(s) are receiving. Similarly, if the congestion is at or near the incoming link, incoming traffic is shed by shedding parent nodes that are sending descriptions in the least important layer of data that the node is receiving. Nodes with available bandwidth are further instructed to subscribe to additional descriptions.

摘要翻译： 公开了一种可以有效地控制拥塞的技术，同时支持网络中的多个计算机之间的流数据的异构性。 多个节点被划分成计算机网络内的多个分配树，其中数据被分成多个优先化层。 当节点遇到数据包丢失时，推断出拥塞的位置。 如果拥塞处于或接近输出链路，则流出流量被减轻，以减轻子节点在子节点正在接收的最不重要的数据层中接收描述的缓冲来减轻拥塞。 类似地，如果拥塞处于或接近传入链路，则通过在发送节点正在接收的最不重要的数据层中发送描述的父节点脱离传入流量。 进一步指示具有可用带宽的节点订阅附加描述。

公开(公告)号：US07543073B2

公开(公告)日：2009-06-02

申请号：US11010040

申请日：2004-12-10

申请人： Philip Chou ,  Anders Klemets ,  Cheng Huang

发明人： Philip Chou ,  Anders Klemets ,  Cheng Huang

IPC分类号： G06F15/16

CPC分类号： H04N21/44004 ,  H04L47/10 ,  H04N21/23406 ,  H04N21/2662 ,  H04N21/6373 ,  H04N21/6379 ,  H04N21/6547

摘要： A system and process for performing an exponentially weighted moving average on streaming data to establish a moving average bit rate of data units is presented. In general, the system or process computes, on a unit-by-unit basis, the product of the moving average bit rate computed for a data unit immediately prior to a unit under consideration and a first fractional weighting factor, added to the product of the instantaneous bit rate of the data unit under consideration and a second fractional weighting factor, wherein at least one fractional weighting factor is not a constant but instead based on the time between data units.

摘要翻译： 提出了一种用于在流数据上执行指数加权移动平均以建立数据单元的移动平均比特率的系统和过程。 一般来说，系统或过程在逐单位的基础上计算紧接在所考虑的单元之前针对数据单元计算的移动平均比特率和第一分数加权因子的乘积， 所考虑的数据单元的瞬时比特率和第二分数加权因子，其中至少一个分数加权因子不是常数，而是基于数据单元之间的时间。

公开(公告)号：US07412376B2

公开(公告)日：2008-08-12

申请号：US10660326

申请日：2003-09-10

申请人： Dinei Florencio ,  Philip Chou

发明人： Dinei Florencio ,  Philip Chou

IPC分类号： G10L11/00 ,  G10L11/02

CPC分类号： G10L25/87 ,  G10L2025/783

摘要： A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.

摘要翻译： “语音起始检测器”提供了可变长度帧缓冲器，与缓冲信号帧的可变传输速率或时间语音压缩相结合。 可变长度缓冲器缓冲在初始分析期间未被清楚地识别为语音或非语音帧的帧。 信号帧的缓冲持续到当前帧被识别为语音或非语音。 如果当前帧被识别为非语音，则缓冲帧被编码为非语音帧。 然而，如果当前帧被识别为语音帧，则搜索缓冲的帧用于语音的实际起始点。 一旦该起始点被识别，则信号以突发方式发送，或者缓冲信号的时间尺度修改被应用于从检测到起始点的帧开始的缓冲帧。 然后将压缩的帧编码为一个或多个语音帧。

公开(公告)号：US20070263607A1

公开(公告)日：2007-11-15

申请号：US11394773

申请日：2006-03-31

申请人： David Milstein ,  David Howell ,  Linda Criddle ,  Michael Malueg ,  Philip Chou

发明人： David Milstein ,  David Howell ,  Linda Criddle ,  Michael Malueg ,  Philip Chou

IPC分类号： H04L12/66

CPC分类号： H04L12/66

摘要： Structured hierarchies for communicating contextual information relating to a VoIP conversation are provided. The structured hierarchies are utilized for efficient communications of various amounts and types of contextual information over a VoIP conversation channel. Information identifying at least one structured hierarchy, which will be used to carry the contextual information, is transmitted during establishment of a conversation between two VoIP enhanced devices and prior to the exchange of contextual information. The structural hierarchy is selected from a set of predefined and declared structured hierarchies. Subsequently transmitted contextual information exchanged between two VoIP enhanced devices is represented in accordance with the identified structural hierarchy. Additionally, the structural hierarchies can be extensible by the addition of more definitions to the current structural hierarchies.

摘要翻译： 提供了用于传送与VoIP会话相关的上下文信息的结构化层级。 结构化层次被用于通过VoIP对话信道有效地通信各种数量和类型的上下文信息。 在两个VoIP增强设备之间的交谈建立之前，以及交换上下文信息之前，发送识别用于携带上下文信息的至少一个结构化层次结构的信息。 从一组预定义和声明的结构化层次结构中选择结构层次结构。 随后根据所识别的结构层次来表示在两个VoIP增强设备之间交换的传送的上下文信息。 此外，通过向当前的结构层次结构添加更多的定义，结构层次结构可以扩展。

公开(公告)号：US20070201371A1

公开(公告)日：2007-08-30

申请号：US11363801

申请日：2006-02-28

申请人： Philip Chou ,  Kamal Jain ,  Dinesh Garg

发明人： Philip Chou ,  Kamal Jain ,  Dinesh Garg

IPC分类号： H04L12/26

CPC分类号： H04L45/123 ,  H04L45/12 ,  H04L45/22 ,  H04L45/28

摘要： Congestion adaptive data routing is leveraged to provide a substantial increase in data throughput in networks with data congestion. By continuously adapting the data routes when a congested route is encountered, the data can reach its destination via alternate routes around the congested area. This is accomplished in a distributed manner where each node provides an alternative path to congestion based on its local knowledge and/or knowledge obtained from neighboring nodes. This allows the data path to be dynamically adjusted for congestion without requiring a centralized body of control. In another instance, data rate changes can be combined with data path changes to increase the efficiency of the data throughput. Alternative routes can be determined based upon the costs associated with selecting that route. Selecting a minimum cost route yields the most efficient transfer of data.

摘要翻译： 拥塞自适应数据路由被用来提供具有数据拥塞的网络中的数据吞吐量的显着增加。 通过在遇到拥塞路由时不断调整数据路由，数据可以通过拥塞区域周围的备用路由到达目的地。 这是以分布式方式实现的，其中每个节点基于从相邻节点获得的本地知识和/或知识提供了拥塞的替代路径。 这允许数据路径被动态地调整以用于拥塞而不需要集中的控制体。 在另一种情况下，数据速率变化可以与数据路径改变相结合，以提高数据吞吐量的效率。 可以基于与选择该路线相关联的成本来确定替代路线。 选择最低成本路线可以最有效地传输数据。

公开(公告)号：US07366172B2

公开(公告)日：2008-04-29

申请号：US11177258

申请日：2005-07-08

申请人： Philip Chou ,  Albert Wang ,  Sanjeev Mehrotra ,  Alexander Mohr

发明人： Philip Chou ,  Albert Wang ,  Sanjeev Mehrotra ,  Alexander Mohr

IPC分类号： H04L12/56

CPC分类号： H04L1/007 ,  H04L1/0009 ,  H04L1/0017 ,  H04L1/06 ,  H04L1/1812 ,  H04L2001/0093

摘要： A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission. A preferred process for generating the error correction streams involves the use of a unique adaptation of the Forward Error Correction (FEC) techniques. This process encodes the transmission data using a linear transform which adds redundant elements. The redundancy permits losses to be corrected because any of the original data elements can be derived from any of the encoded elements. Thus, as long as enough of the encoded data elements are received so as to equal the number of the original data elements, it is possible to derive all the original elements.

摘要翻译： 一种用于在异构分组网络（例如因特网）下校正在实时媒体的接收机驱动分层多播（RLM）期间发生的错误和损失的系统和方法。 这是通过用一层或多层纠错信息增强RLM来实现的。 这允许每个接收机通过订阅至少一个纠错层来分别优化所接收的音频和视频信息的质量。 理想地，RLM中的每个源层将具有与其相关联的一个或多个多播的纠错数据流（即，层）。 每个纠错层将包含可用于替换相关源层丢失的分组的信息。 提出了多于一个纠错层，因为包含在替换相关源流中丢失的分组所需的数据流中的一些纠错分组本身可能在传输中丢失。 用于产生纠错流的优选过程涉及使用前向纠错（FEC）技术的唯一适配。 该过程使用添加冗余元素的线性变换对传输数据进行编码。 冗余允许修正损失，因为任何原始数据元素可以从任何编码元素导出。 因此，只要接收到足够的编码数据元素以便等于原始数据元素的数量，就有可能导出所有的原始元素。

公开(公告)号：US20070253407A1

公开(公告)日：2007-11-01

申请号：US11415323

申请日：2006-05-01

申请人： Kaunsan Wang ,  Michael Malueg ,  Philip Chou

发明人： Kaunsan Wang ,  Michael Malueg ,  Philip Chou

IPC分类号： H04L12/66

CPC分类号： H04L12/66 ,  H04L29/06027 ,  H04L65/1069 ,  H04L65/607 ,  H04L65/80

摘要： A method and system for providing enhanced VoIP services relating to the use of callee rules and/or caller rules is provided. A callee may specify callee rules defining the callee preferences such as which VoIP device of the callee is appropriate for responding to an incoming communication from a specified caller. The callee rules may define a priority of VoIP devices of the callee, designating in which order the VoIP devices are to be notified of any incoming communication from a specified caller. Similarly, a caller can specify caller rules defining the caller preferences. The method and system compares the callee rules and the caller rules to establish a communication channel. As such, various enhanced VoIP services can be tailored based on the callee rules and the caller rules.

摘要翻译： 提供了一种用于提供与使用被叫方规则和/或呼叫者规则有关的增强的VoIP服务的方法和系统。 被叫方可以指定定义被叫用户偏好的被叫方规则，例如被叫方的哪个VoIP设备适合于响应来自指定呼叫者的传入通信。 受理者规则可以定义被叫方的VoIP设备的优先级，指定VoIP设备将从哪个命令通知来自指定呼叫者的任何传入通信。 类似地，呼叫者可以指定定义呼叫者偏好的呼叫者规则。 该方法和系统比较被叫规则和呼叫者规则以建立通信信道。 因此，可以基于被叫规则和呼叫者规则来定制各种增强的VoIP服务。