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1.发明授权Method and apparatus to reduce jitter and end-to-end delay for multimedia data signalling 失效用于减少多媒体数据信令的抖动和端到端延迟的方法和装置
公开(公告)号：US5996018A
公开(公告)日：1999-11-30
申请号：US758071
申请日：1996-11-27
CPC分类号： H04J3/0632 , G06F13/387
摘要： A method and an apparatus for reducing the jitter and end-to-end delay on lines of a packet switching network conveying voice or video digitalized data for one or more connections between a local source and a remote source at a constant bit rate.The method and apparatus of the invention are for use in a voice or video processor of a voice or video processing server of a network node; the method and the apparatus provide a way of controlling the remote traffic rate from the remote source before accessing the processor without using an external clocking such as the network clock.The solution proposed by the invention consists in buffering the remote and local traffics to adapt the remote traffic rate to the local traffic rate, which is supposed having a limited jitter, while sequencing of the access of the buffered data to the processor.
摘要翻译： 一种用于减少分组交换网络的线路上的抖动和端到端延迟的方法和装置，其以恒定的比特率传送本地源和远程源之间的一个或多个连接的语音或视频数字化数据。 本发明的方法和装置用于网络节点的语音或视频处理服务器的语音或视频处理器; 该方法和装置提供了在访问处理器之前控制来自远程源的远程业务速率的方式，而不使用诸如网络时钟的外部时钟。 本发明提出的解决方案在于缓存远程和本地业务，以使远程业务速率适应本地业务速率，其被认为具有有限的抖动，同时缓冲数据到处理器的访问排序。
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公开(公告)号：US06912224B1
公开(公告)日：2005-06-28
申请号：US09021707
申请日：1998-02-10
申请人： Maurice Duault , Bernard Pucci , Serge Leroy , Gerard Richter , Francois Kermarec , Aline Fichou
发明人： Maurice Duault , Bernard Pucci , Serge Leroy , Gerard Richter , Francois Kermarec , Aline Fichou
CPC分类号： H04L49/90
摘要： An adaptive apparatus and method for managing a playout buffer (POB) in an edge node of a packet-based data communications network, such as an ATM network, in order to reduce the end-to-end communication delay introduced thereby and to allow the POB filling level to be corrected in the event of clock speed differences. A monitoring mechanism determines if the minimum filling level of the playout buffer over a time period whereby the average filling level of the playout buffer is reduced according to the minimum filling level of the playout buffer over the time period.
摘要翻译： 一种用于管理诸如ATM网络的基于分组的数据通信网络的边缘节点中的播出缓冲器（POB）的自适应装置和方法，以便减少由此引入的端到端通信延迟并允许 在时钟速度差异的情况下要纠正POB填充级别。 监视机构确定在一段时间内播放缓冲器的最小填充水平是否根据播放缓冲器在该时间段内的最小填充水平而减小播放缓冲器的平均填充水平。
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3.发明授权Data processing method for efficiently transporting multimedia packets over a conventional digital packet switching network 失效用于通过传统数字分组交换网络高效传输多媒体分组的数据处理方法
公开(公告)号：US5930265A
公开(公告)日：1999-07-27
申请号：US782694
申请日：1997-01-16
申请人： Maurice Duault , Laurent Nicolas , Claude Galand , Bernard Pucci
发明人： Maurice Duault , Laurent Nicolas , Claude Galand , Bernard Pucci
CPC分类号： H04Q11/0478 , H04L2012/5652
摘要： A data processing method for efficiently transporting multimedia data packets of fixed and/or variable length over an Assynchronous Transfer Mode (ATM) network made to transport fixed length ATM cells including a fixed length user data payload and a fixed length ATM header. The data processing method includes concatenating said fixed and/or variable length user data and appending said concatenated data with a so-called trailer defining the various concatenated user data lengths and identifications, for being further split into ATM cells payloads before being transmitted over said ATM network.
摘要翻译： 一种数据处理方法，用于通过异步传输模式（ATM）网络高效地传输固定和/或可变长度的多媒体数据分组，用于传输固定长度的ATM信元，包括固定长度用户数据有效载荷和固定长度的ATM报头。 数据处理方法包括连接所述固定和/或可变长度的用户数据，并将所述连接的数据与定义各种级联用户数据长度和标识的所谓拖尾相连，以便在通过所述ATM发送之前进一步分割成ATM信元有效载荷 网络。
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4.发明授权Method and system for monitoring traffic to optimize the bandwidth reserved to an audio channel connection in a high speed digital network 失效用于监测流量以优化在高速数字网络中为音频信道连接预留的带宽的方法和系统
公开(公告)号：US5912894A
公开(公告)日：1999-06-15
申请号：US786915
申请日：1997-01-22
申请人： Maurice Duault , Claude Galand , Gerard Lebizay , Bernard Pucci , Gerard Richter
发明人： Maurice Duault , Claude Galand , Gerard Lebizay , Bernard Pucci , Gerard Richter
IPC分类号： H04L12/54 , H04L12/801 , H04L12/853 , H04L12/911 , H04L12/913 , H04L12/927 , H04M3/22 , H04M7/12 , H04J3/17
CPC分类号： H04L12/5695 , H04L47/15 , H04L47/2416 , H04L47/724 , H04L47/801 , H04L47/829 , H04M3/2254 , H04M7/12
摘要： This invention deals with a method and system for dynamically adjusting the communication bandwidth assigned to an audio channel connection in a high speed digital network. More particularly, the invention is made to track the activity of a voice assigned connection (e.g. a PBX or PABX entry to the network), define a so-called activity bit for each block of audio channel signal and then dynamically adjust the assigned network communication bandwidth accordingly. Adjustment of the communication bandwidth is accomplished by integrating the audio channel activity bits through a predefined integration function, the result of which is then compared against predefined threshold values in order to determine an appropriate bandwidth setting for the audio channel.
摘要翻译： 本发明涉及用于动态调整分配给高速数字网络中的音频信道连接的通信带宽的方法和系统。 更具体地说，本发明用于追踪语音分配的连接（例如，到网络的PBX或PABX条目）的活动，为每个音频信道信号块定义所谓的活动比特，然后动态地调整分配的网络通信 带宽相应。 通过将音频信道活动比特整合到预定义的积分函数来实现通信带宽的调整，然后将其结果与预定义的阈值进行比较，以便为音频信道确定适当的带宽设置。
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公开(公告)号：US5600641A
公开(公告)日：1997-02-04
申请号：US451580
申请日：1995-05-26
申请人： Maurice Duault , Gerard Lebizay , Claude Galand
发明人： Maurice Duault , Gerard Lebizay , Claude Galand
CPC分类号： H04L49/3081 , H04L12/6418 , H04Q11/0478 , H04J2203/0087 , H04L2012/562 , H04L2012/5626 , H04L2012/563 , H04L2012/5638 , H04L2012/5651 , H04L2012/6464 , H04L2012/6481 , H04L2012/6489
摘要： Voice circuit emulation system in a packet switching network includes a plurality of switching nodes (SW-1, SW-2, . . . ) interconnected by connection lines (TL-1, TL-2, . . . ), enabling voice signals to be transmitted in circuit emulation packets from any source exchange telephone device (PBX-A) to any destination exchange telephone device (PBX-B). Each switching node (SW-1 or SW-2) includes a circuit emulation server (CES-1 or CES-2) including a plurality of connection tables corresponding each to an incoming connection line, each table containing for each incoming connection line, the identification of each outgoing connection line associated to each slot of the circuit emulation packet received from the incoming connection line and the identification of each slot of the packet to be transmitted on the outgoing connection line. A switching module looks up the connection table for each slot contained in each incoming circuit emulation packet received from all incoming lines and thereby transferring the contents thereof to the slot of the outgoing connection line identified in said connection table.
摘要翻译： 分组交换网络中的语音电路仿真系统包括通过连接线（TL-1，TL-2 ...）互连的多个交换节点（SW-1，SW-2 ...），使得语音信号 在电路仿真包中从任何源交换电话设备（PBX-A）发送到任何目的地交换电话设备（PBX-B）。 每个交换节点（SW-1或SW-2）包括电路仿真服务器（CES-1或CES-2），该电路仿真服务器（CES-1或CES-2）包括与输入连接线对应的多个连接表，每个表包含每个输入连接线， 识别与从输入连接线接收的电路仿真分组的每个时隙相关联的每个输出连接线以及要在输出连接线上发送的分组的每个时隙的标识。 切换模块查找从所有输入线路接收到的每个输入电路仿真分组中包含的每个时隙的连接表，从而将其内容传送到在所述连接表中标识的输出连接线的时隙。
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公开(公告)号：US06226266B1
公开(公告)日：2001-05-01
申请号：US08946237
申请日：1997-10-07
申请人： Claude Galand , Maurice Duault , Aline Fichou , Olivier Bertin
发明人： Claude Galand , Maurice Duault , Aline Fichou , Olivier Bertin
IPC分类号： H04L1256
CPC分类号： H04Q11/0478 , H04L2012/5671 , H04L2012/5684
摘要： An apparatus and method that determines the end-to-end transit delay at each node of a path, in accordance with the selected probability value indicative of the probability to experience a delay at each node that is smaller than the computed transit delay. Then the computed transit delays per nodes are combined to obtain the end-to-end delay of the path, the combination being either an arithmetic operation or a convolution operation. A method to approximate the convolution operation is also disclosed.
摘要翻译： 根据所选择的概率值，其指示在每个节点处经历比所计算的过渡延迟小的延迟的概率，确定路径的每个节点处的端到端传输延迟的装置和方法。 然后将每个节点的计算的传输延迟组合以获得路径的端到端延迟，该组合是算术运算或卷积运算。 还公开了近似卷积运算的方法。
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公开(公告)号：US06529512B1
公开(公告)日：2003-03-04
申请号：US09031755
申请日：1998-02-27
IPC分类号： H04Q1100
CPC分类号： H04Q11/0414 , H04Q2213/13034 , H04Q2213/13056 , H04Q2213/13174 , H04Q2213/13204 , H04Q2213/13216 , H04Q2213/1329 , H04Q2213/13292 , H04Q2213/13389
摘要： A communication system and method for compressing data in a transmission system wherein multiplexed channels are transported over a transmission network of the type comprising a plurality of switching nodes interconnected by connection lines, the exchange of data signals carried out by switching the channels in the network between two exchange telephone devices, and each of the multiplexed channels transporting data bytes representing the data signals from one source exchange telephone device to one destination exchange device during an exchange of information therebetween through the intermediary of a compression/decompression device. The method comprises the steps of comparing, for each multiplexed channel, the signal value associated to each one of a plurality of “n” consecutive data bytes to a predetermined threshold; deleting, in case said signal value for all said “n” data bytes is less than the predetermined threshold, all bits which are not necessary to represent the signal value from each of the “n” data bytes; building a compression frame by concatenating either the “n” data bytes when they are not modified or the “n” modified data bytes when bits have been deleted therefrom, and adding to each of said groups an identifier indicating whether said data bytes are modified or not before transmitting said compression frame over said transmission network. Decompressing the frame by determining the identifier value indicating the composition of the bytes; removing the identifier from the bytes; loading the bytes into a buffer and transmitting the bytes to a destination exchange telephone device.
摘要翻译： 一种用于在传输系统中压缩数据的通信系统和方法，其中多路复用信道通过包括由连接线互连的多个交换节点的类型的传输网络传送，通过在网络中切换网络中的信道而进行的数据信号的交换 两个交换电话设备，并且每个复用信道在通过压缩/解压缩设备的中间交换信息期间将表示数据信号的数据字节从一个源交换电话设备传送到一个目的地交换设备。 该方法包括以下步骤：对于每个复用的信道，将与多个“n”个连续数据字节中的每一个相关联的信号值与预定阈值进行比较; 删除在所有所述“n”个数据字节的所述信号值小于预定阈值的情况下，从“n”个数据字节中的每个数据字节中不需要表示信号值的所有位; 通过在“n”数据字节未被修改时连接“n”个数据字节，或者当从其中删除位时，将“n”个修改的数据字节连接起来构建压缩帧，并向每个所述组添加指示所述数据字节是否被修改的标识符， 而不是在通过所述传输网络发送所述压缩帧之前。 通过确定指示字节的组成的标识符值来解压缩帧; 从字节中删除标识符; 将字节加载到缓冲器中并将字节传送到目的地交换电话设备。
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公开(公告)号：US5638365A
公开(公告)日：1997-06-10
申请号：US526344
申请日：1995-09-11
申请人： Maurice Duault , Claude Galand
发明人： Maurice Duault , Claude Galand
CPC分类号： H04L12/5602 , H04Q11/0478 , H04L2012/563 , H04L2012/5632 , H04L2012/5634 , H04L2012/5652
摘要： A method dynamically changes the bit rate or bandwidth of constant bit rate data structures in an Asynchronous Transfer Mode (ATM) communications environment. The method defines within a data channel a Change indicator (CI) indicative of an end user's request for a bit rate change within said data structures. The Change Indicator is continuously transmitted from a source side to a destination side in the ATM environment, along with the data structures on said data channel. Upon receipt at the destination side of a user's request of a bit rate change, the destination side modifies the value of said Change Indicator. The source side, acknowledges the modification of the value of the Change Indicator, whereby the transmission of data structures continues on the data channel with a new constant bit rate.
摘要翻译： 一种在异步传输模式（ATM）通信环境中动态地改变恒定比特率数据结构的比特率或带宽的方法。 该方法在数据信道内定义指示最终用户对所述数据结构内的比特率改变的请求的改变指示符（CI）。 变化指示器连同在ATM环境中的源侧到目的地一起连同数据通道上的数据结构一起发送。 在用户对比特率改变的请求的目的地侧接收到目的地侧修改所述改变指示符的值。 源端确认修改变量指示器的值，从而数据结构的传输在数据通道上以新的恒定位速率继续。
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9.发明授权
公开(公告)号：US6144658A
公开(公告)日：2000-11-07
申请号：US989585
申请日：1997-12-12
申请人： Gerard Lebizay , Maurice Duault , Bernard Pucci , Gerard Richter
发明人： Gerard Lebizay , Maurice Duault , Bernard Pucci , Gerard Richter
CPC分类号： H04L12/6418 , H04J3/17 , H04L2012/6467 , H04L2012/6481
摘要： Repetitive packets in a voice/data stream, are detected and suppressed in the transmitting side of a network, after a predefined number of consecutive repetitive packets have been transmitted. Then, at the receiving side of the network, suppressed repetitive packets are reconstituted by filling the resulting gap in the voice/data stream with the repetitive pattern contained in the last received repetitive packet. When the input voice/data stream is compressed, only non-repetitive packets are compressed so that repetitive patterns are not corrupted by compression.
摘要翻译： 在发送预定数量的连续重复分组之后，在网络的发送侧检测并抑制语音/数据流中的重复分组。 然后，在网络的接收侧，通过用包含在最后接收的重复分组中的重复模式填充语音/数据流中的所得到的间隙来重构抑制的重复分组。 当输入的语音/数据流被压缩时，只有非重复的数据包被压缩，使得重复的模式不被压缩破坏。
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10.发明授权Method and a system for silence removal in a voice signal transported through a communication network 失效通过通信网络传输的语音信号中的静音消除方法和系统
公开(公告)号：US5870397A
公开(公告)日：1999-02-09
申请号：US695280
申请日：1996-08-06
IPC分类号： G10L19/012 , H04B1/46 , H04L12/70 , H04Q11/04 , H04T3/17
CPC分类号： G10L19/012 , H04Q11/0478 , H04B1/46 , H04L2012/5616 , H04L2012/5647 , H04L2012/5652 , H04L2012/5671 , H04Q2213/13034 , H04Q2213/13056 , H04Q2213/13166 , H04Q2213/13201 , H04Q2213/13216 , H04Q2213/1329
摘要： A method and an apparatus for removing the silence from the digitalized voice signals conveyed through packets or cells switching networks. The silence samples are neither packetized nor sent over the network but are regenerated at the output of the network. The silence samples generated are white noise samples, where the level is adapted to the background noise of the silence samples received at the input node of the network. For long periods of silence, the white noise level is periodically refreshed to be adapted to the last silence samples received at the input node of the network. The method provides also a control of packet or cell loss. The method uses are not control packets; in the later case, it can be used for ATM networks with AAL1. The method is implemented as a program executed in a Digital Signal Processor located on adapter cards dedicated to voice processing in the network access nodes.
摘要翻译： 一种用于从通过分组或小区交换网络传送的数字化语音信号中去除静音的方法和装置。 沉默样本既不打包也不通过网络发送，而是在网络的输出端重新生成。 产生的静音样本是白噪声样本，其中该电平适应于在网络的输入节点接收的静音样本的背景噪声。 对于长时间的静音，白噪声电平被周期性刷新以适应于在网络的输入节点处接收到的最后沉默样本。 该方法还提供了分组或信元丢失的控制。 该方法使用的不是控制包; 在后一种情况下，它可以用于具有AAL1的ATM网络。 该方法被实现为位于专用于网络接入节点中的语音处理的适配卡上的数字信号处理器中执行的程序。
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