End-to-end delay estimation in high speed communication networks
    31.
    发明授权
    End-to-end delay estimation in high speed communication networks 失效
    高速通信网络中的端到端延迟估计

    公开(公告)号:US06226266B1

    公开(公告)日:2001-05-01

    申请号:US08946237

    申请日:1997-10-07

    IPC分类号: H04L1256

    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。

    Method and system for optimizing the connection set up time in high
speed communication networks for recovering from network failure
    32.
    发明授权
    Method and system for optimizing the connection set up time in high speed communication networks for recovering from network failure 失效
    在高速通信网络中优化连接建立时间以从网络故障中恢复的方法和系统

    公开(公告)号:US6038212A

    公开(公告)日:2000-03-14

    申请号:US946243

    申请日:1997-10-07

    摘要: This method enables optimizing the time required for reestablishing connections between end users attached to a data communication network, which connections were disrupted due to a network failure. The network includes access nodes and transit nodes interconnected with network links/trunks (with no specific distinction being herein required between both designations of a communication line). The end users are attached to the network through access nodes and each said access node permanently stores an image of the current network trunk including the number N.sub.i of connections currently supported by said network trunk. Upon detection of a trunk failure, each access node supporting connections affected by said failure, is made aware of the total number (N.sub.i) of connections in each priority group affected by the failure, together with a network dependent parameter (TP) representing the elementary processing time required to reroute a single network connection. Then each access node may independently, start a first reconnection set-up procedure at a time Ri randomly selected between zero and (N.sub.i -n.sub.i) *TP, where n.sub.i is the number of connections supported by the access node in each priority group, and then space the required subsequent reconnections set-ups by a time equal to (T.sub.i -R.sub.i)/n.sub.i, with T.sub.i =N.sub.i *TP.

    摘要翻译: 该方法能够优化重新建立连接到数据通信网络的终端用户之间的连接所需的时间,哪些连接由于网络故障而中断。 网络包括与网络链路/中继线互连的接入节点和传输节点(在通信线路的两个指定之间,这里不需要具体区分)。 最终用户通过接入节点连接到网络,并且每个所述接入节点永久地存储包括当前由所述网络中继线支持的连接的数目Ni的当前网络中继的图像。 在检测到中继线故障时,支持受所述故障影响的连接的每个接入节点被识别出由故障影响的每个优先级组中的连接的总数(Ni),以及表示基站的网络相关参数(TP) 重新路由单个网络连接所需的处理时间。 然后,每个接入节点可以独立地在零和(Ni-ni)* TP之间随机选择的时间Ri开始第一重新连接建立过程,其中,ni是每个优先级组中的接入节点支持的连接数,以及 然后将所需的后续重新连接设置空间等于(Ti-Ri)/ ni,Ti = Ni * TP。

    Method and system for non-disruptively assigning link bandwidth to a
user in a high speed digital network
    33.
    发明授权
    Method and system for non-disruptively assigning link bandwidth to a user in a high speed digital network 失效
    用于在高速数字网络中不间断地将链路带宽分配给用户的方法和系统

    公开(公告)号:US5881050A

    公开(公告)日:1999-03-09

    申请号:US785944

    申请日:1997-01-22

    摘要: A method based on predefined connection priorities for assigning link bandwidth to a requesting user in a high speed digital network interconnecting network users through a path including network nodes connected through high speed links. According to this method, a predefined reservable link bandwidth is split into so-called nominal bandwidth portions and common bandwidth portions, both assignable to the same connections on a priority basis. Each of the common bandwidth priorities is individually related to a nominal bandwidth priority through a predefined relationship, making the common bandwidth priorities always lower than any nominal priority. In this way the requested link connection bandwidth, whatever be its nominal priority, is made preemptable primarily on all common bandwidth, thus avoiding the disruption of any network connection which is already established.

    摘要翻译: 一种基于预定义的连接优先级的方法,用于将高速数字网络中的请求用户分配链路带宽,通过包括通过高速链路连接的网络节点的路径互连网络用户。 根据该方法,将预定义的可预留链路带宽分为所谓的标称带宽部分和公共带宽部分,两者都可优先分配给相同的连接。 每个公共带宽优先级通过预定义的关系单独地与标称带宽优先级相关,使得共同带宽优先级始终低于任何标称优先级。 以这种方式,无论作为其标称优先级,所请求的链路连接带宽主要在所有公共带宽上可抢占,从而避免已经建立的任何网络连接的中断。

    Method and apparatus for DTMF detection
    34.
    发明授权
    Method and apparatus for DTMF detection 失效
    用于DTMF检测的方法和装置

    公开(公告)号:US5353345A

    公开(公告)日:1994-10-04

    申请号:US960744

    申请日:1992-10-14

    申请人: Claude Galand

    发明人: Claude Galand

    IPC分类号: H04Q1/457 H04Q1/46 H04M3/00

    CPC分类号: H04Q1/46 H04Q1/4575

    摘要: A DTMF tone is detected through the tracking of two tunable filters HAVING coefficients within so-called tunnels each limited by thresholds derived from the tone frequency components to be detected. In and out-band energies E1(n) and E2(n) are used to further validate tone detection.

    摘要翻译: 通过跟踪两个可调滤波器来检测DTMF音,这些滤波器在所谓的隧道内具有各自受限于由要检测的音调频率分量导出的阈值的系数。 输入和输出带外能量E1(n)和E2(n)用于进一步验证音调检测。

    Voice coding process and device for implementing said process
    35.
    发明授权
    Voice coding process and device for implementing said process 失效
    语音编码过程和用于实现所述过程的设备

    公开(公告)号:US5001758A

    公开(公告)日:1991-03-19

    申请号:US35806

    申请日:1987-04-08

    IPC分类号: H04B14/04 G10L19/06 H03M7/30

    CPC分类号: G10L19/06

    摘要: The voice signal is analyzed to derive therefrom a low frequency base band signal, linear prediction coefficients and high frequency (HF) descriptors. Said HF descriptors include HF energy indications as well as indications relative to the phase shift between the low frequency and the high frequency band. Said HF descriptors are used during the voice synthesis operation to provide an inphase HF bandwidth component to be added to the base band prior to be used for driving a linear prediction synthesis filter tuned using said linear prediction parameters.

    Method of reconstructing lost data in a digital voice transmission
system and transmission system using said method
    37.
    发明授权
    Method of reconstructing lost data in a digital voice transmission system and transmission system using said method 失效
    使用所述方法在数字语音传输系统和传输系统中重建丢失数据的方法

    公开(公告)号:US4907277A

    公开(公告)日:1990-03-06

    申请号:US212121

    申请日:1988-06-27

    摘要: A method of reconstructing, at the receiving end, digital data defining an encoded voice signal segment lost in transmission between the transmitter and the receiver in a transmission system wherein a low-frequency signal such as a residual baseband signal is derived at the transmitting end from the signal to be encoded and is then distributed among several sub-bands whose sampled contents are quantized separately. The reconstruction method includes a step of analyzing the received signal to detect any missing segment thereof and, as the case may be, to initiate an analysis, within each sub-band, of the segment(s) adjacent to the lost segment, so as to generate a term relating to the period T.sub.(k) of the signal so analyzed and to reconstruct a segment of signal of period T.sub.(k) intended to be substituted for the lost segment. Cyclic redundancy is used to detect errors, and modulo sequence numbering identification is used to detect loss of packet.

    摘要翻译: 在接收端重建在传输系统中定义在发射机和接收机之间传输中丢失的编码语音信号段的数字数据的方法,其中诸如残留基带信号的低频信号在发送端从 要编码的信号然后分配在分别量化其采样内容的几个子带之间。 重建方法包括分析接收到的信号以检测其丢失的段,并且根据情况在每个子带内发起与丢失段相邻的段的分析,以便 以产生与所分析的信号的周期T(k)有关的术语,并重建期望替代丢失段的周期T(k)的信号段。 循环冗余用于检测错误,模数序列识别用于检测数据包的丢失。

    Multirate digital transmission method and device for implementing said
method
    38.
    发明授权
    Multirate digital transmission method and device for implementing said method 失效
    用于实现所述方法的多速率数字传输方法和装置

    公开(公告)号:US4790015A

    公开(公告)日:1988-12-06

    申请号:US31152

    申请日:1987-03-25

    摘要: This coder dispatches the bits resulting from the coding operation of a speech signal into a format for transmission at a rate chosen from a plurality of predetermined transmission rates. More specifically, the contents of at least part of the speech signal frequency bandwidth is split into several sub-bands. Said sub-bands are regrouped into sub-groups, each of said sub-groups corresponding to at least one of said possible transmission rates, i.e. to at least one sub-group coding bit rate. The signal samples belonging to each sub-group are recoded through a dynamical allocation of requantizing steps between the sub-bands. The obtained bits resulting from the recoding are dispatched into a multirate frame according to the sub-group which they belong to.

    摘要翻译: 该编码器将从语音信号的编码操作产生的比特分派成从多个预定传输速率选择的速率进行传输的格式。 更具体地,将语音信号频带宽度的至少一部分的内容分成几个子带。 所述子带被重新分组为子组,每个所述子组对应于所述可能传输速率中的至少一个,即至少一个子组编码比特率。 属于每个子组的信号样本通过在子带之间的重新调整步骤的动态分配被重新编码。 根据它们属于的子组,将从重新编码产生的获得的比特分派到多速率帧。

    Speech coding method and device for implementing the improved method
    39.
    发明授权
    Speech coding method and device for implementing the improved method 失效
    用于实现改进方法的语音编码方法和装置

    公开(公告)号:US4464783A

    公开(公告)日:1984-08-07

    申请号:US369997

    申请日:1982-04-20

    CPC分类号: H04B1/667

    摘要: This improved speech signal Block Coded PCM (BCPCM) system reduces the number of bits allocated to transmitting the scale factor, thereby releasing bits for allocation to coding samples in the associated block of samples. The scale factor (c) is calculated for every 16 millisecond block of samples. However, the scale factor will be transmitted only once per 32 millisecond block if there is no significant difference between the two sequential values. The original speech signal is split into 16 frequency subbands, each subband initials sampled and 12-bit coded, then requantized in BCPCM at dynamically variable bit rates depending on the scale factor transmission rate.

    摘要翻译: 这种改进的语音信号块编码PCM(BCPCM)系统减少了分配给发送比例因子的比特数,从而释放用于分配给相关采样块中的编码样本的比特。 每16毫秒样本块计算比例因子(c)。 但是,如果两个顺序值之间没有显着差异,则缩放因子将仅在每32毫秒块传输一次。 原始语音信号分为16个频率子带,每个子带采样和12位编码,然后根据比例因子传输速率以动态可变比特率在BCPCM中重新量化。