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    • 1. 发明授权
    • Method and system for monitoring and management of the performance of real-time networks
    • 监控和管理实时网络性能的方法和系统
    • US07453815B1
    • 2008-11-18
    • US10735982
    • 2003-12-15
    • David A. GrabelskyIkhlaq S. SidhuGuido M. SchusterJacek A. Grabiec
    • David A. GrabelskyIkhlaq S. SidhuGuido M. SchusterJacek A. Grabiec
    • H04J3/14H04L12/26
    • H04L43/067H04L43/06H04L43/08
    • Gateway routers for real-time networks have the ability to collect delay, loss, and jitter statistics on a per-connection basis. It is possible to use this information not only to monitor the quality of individual voice calls and other real-time connections, but also to evaluate the overall performance of the underlying network. This paper describes a method for monitoring and managing the performance of a real-time data network that supports voice, video and other real-time services. In the described embodiments, the RTCP mechanisms of RTP for sender and receiver reporting be used to relay performance information to one or more network monitoring sites for analysis and interpretation. In addition, gateway routers are organized and managed within a hierarchy that allows the monitoring function to localize it view of network conditions within the hierarchy; and the monitoring of network performance can occur on various time scales.
    • 用于实时网络的网关路由器可以在每个连接的基础上收集延迟,丢失和抖动统计信息。 可以使用此信息不仅可以监视单个语音通话和其他实时连接的质量,还可以评估底层网络的整体性能。 本文介绍了一种用于监视和管理支持语音,视频和其他实时业务的实时数据网络性能的方法。 在所描述的实施例中,用于发送者和接收者报告的RTP的RTCP机制用于将性能信息中继到一个或多个网络监控站点用于分析和解释。 此外,网关路由器在层次结构内组织和管理,允许监视功能将其视图中的网络条件定位在层次结构中; 网络性能的监控可以在不同的时间尺度上进行。
    • 2. 发明授权
    • System and method for providing telephone service having private branch exchange features in a voice-over-data network telephony system
    • 用于在数据网络话音系统中提供具有专用分支交换特征的电话业务的系统和方法
    • US06804224B1
    • 2004-10-12
    • US09515365
    • 2000-02-29
    • Guido M. SchusterIkhlaq S. SidhuJerry J. MahlerFrederick D. DeanJacek A. Grabiec
    • Guido M. SchusterIkhlaq S. SidhuJerry J. MahlerFrederick D. DeanJacek A. Grabiec
    • H04L1266
    • H04M7/1285H04M7/1245
    • A system and method for providing telephone service to a user of a telecommunications device using a data network service provider. The data network service provider has a local service host that is accessible by a local access identifier. A caller uses a telecommunications device to dial the local access identifier to connect to the local service host. In response to a prompt, the caller dials a telephone extension that identifies the callee's telecommunications device. The local service host receives the telephone extension and verifies that the callee is a subscriber. The local service host then retrieves the gateway nearest the callee telecommunications device and opens a voice-over-data channel between the callee and caller gateways. The telephone conversation then proceeds between the callee and caller telecommunications devices over a public switched telephone network connection to the caller gateway, the voice-over-data channel and the PSTN connection to the callee telecommunications device.
    • 一种用于使用数据网络服务提供商向电信设备的用户提供电话服务的系统和方法。 数据网络服务提供商具有可由本地访问标识符访问的本地服务主机。 呼叫者使用电信设备拨打本地接入标识符来连接本地业务主机。 响应于提示,呼叫者拨打一个电话分机,标识被叫方的电信设备。 本地服务主机接收电话分机,并验证被叫方是否为用户。 本地服务主机然后检索最近被叫电信设备的网关,并在被叫方和呼叫者网关之间打开数据通道。 电话会话然后通过公共交换电话网连接到呼叫者网关,语音数据信道和到被叫电信设备的PSTN连接,在被叫方和呼叫者电信设备之间进行。
    • 4. 发明授权
    • Forward error correction system for packet based real time media
    • 用于基于分组的实时媒体的前向纠错系统
    • US06487690B1
    • 2002-11-26
    • US09707567
    • 2000-11-06
    • Guido M. SchusterJerry MahlerIkhlaq SidhuMichael Borella
    • Guido M. SchusterJerry MahlerIkhlaq SidhuMichael Borella
    • H03M1300
    • H04L1/008H04L1/0057
    • A computationally simple yet powerful forward error correction code scheme for transmission of real-time media signals, such as digitized voice, video or audio, in a packet switched network such as the Internet. An encoder at the sending end derives p redundancy blocks from each group of a k payload blocks and concatenates the redundancy blocks, respectively, with payload blocks in the next group of k payload blocks. At the receiving end, a decoder may recover up to p missing packets in a group of k packets, provided with the p redundancy blocks carried by the next group of k packets. The invention thereby enables correction from the loss of multiple packets in a row, without significantly increasing the data rate or otherwise delaying transmission.
    • 一种用于在诸如因特网的分组交换网络中传输诸如数字化语音,视频或音频的实时媒体信号的计算简单而强大的前向纠错码方案。 发送端的编码器从k个有效载荷块的每组中导出p个冗余块,并将冗余块分别与下一组k个有效载荷块中的有效负载块相连。 在接收端,解码器可以恢复一组k个分组中的p个丢失的分组,该分组具有由下一组k个分组携带的p个冗余块。 因此,本发明能够从一行中的多个分组的丢失中进行校正,而不会显着增加数据速率或以其他方式延迟传输。
    • 5. 发明授权
    • Method and apparatus for real time communication system buffer size and error correction coding selection
    • 用于实时通信系统缓冲器大小和纠错编码选择的方法和装置
    • US06366959B1
    • 2002-04-02
    • US09322836
    • 1999-05-28
    • Ikhlaq S. SidhuGuido M. SchusterJames M. Kroll
    • Ikhlaq S. SidhuGuido M. SchusterJames M. Kroll
    • G06F1300
    • H04J3/0632H04J3/0638H04J3/0644H04J3/0682H04L12/6418H04L29/06027H04L49/206H04L49/25H04L49/253H04L49/30H04L49/557H04L65/604H04L65/80H04L2012/5616H04L2012/5647H04L2012/5649H04L2012/5671H04L2012/5674H04L2012/5681H04L2012/6429H04L2012/6472H04L2012/6475H04L2012/6481H04L2012/6483H04L2012/6489H04Q11/0478
    • A method and apparatus for communication system buffer size and error correction coding selection. A method includes the steps of receiving a stream of data packets by a real time receiver that includes a buffer management device, a first plurality of jitter buffers, and a second plurality jitter buffers. The first and second plurality of jitter buffers are evaluated and a first and a second optimal jitter buffer is chosen. The first and the second optimal jitter buffer has an associated conditional optimal performance characteristic. The conditional characteristics are compared and a preferred buffer of the receiver is selected. The apparatus includes a receiving device including a first set of jitter buffers and a second set of jitter buffers with error coding. The first set includes a plurality of buffers and a second plurality of buffers maintained in the second set of buffers. The apparatus also includes a means for comparing the first plurality of buffers and the second plurality of buffers, a means for selecting a first optimal buffer from the first plurality of buffers, and a means for selecting a second optimal buffer from the second plurality of buffers. Either the first or the second selected optimal decoder determines the receiver buffer size and whether forward error correction is utilized.
    • 一种用于通信系统缓冲器大小和纠错编码选择的方法和装置。 一种方法包括以下步骤:由包括缓冲器管理装置,第一多个抖动缓冲器和第二多个抖动缓冲器的实时接收器接收数据分组流。 评估第一和第二多个抖动缓冲器,并且选择第一和第二最佳抖动缓冲器。 第一和第二最佳抖动缓冲器具有相关联的条件最优性能特征。 比较条件特征并选择接收机的优选缓冲器。 该装置包括接收装置,其包括第一组抖动缓冲器和具有错误编码的第二组抖动缓冲器。 第一组包括多个缓冲器和第二组缓冲器,第二组缓冲器保持在第二组缓冲器中。 该装置还包括用于比较第一多个缓冲器和第二多个缓冲器的装置,用于从第一多个缓冲器中选择第一最佳缓冲器的装置,以及用于从第二多个缓冲器中选择第二最佳缓冲器的装置 。 第一或第二选择的最佳解码器确定接收器缓冲器大小以及是否利用前向纠错。
    • 7. 发明授权
    • System and method for selecting a loudest speaker by comparing average
frame gains
    • 通过比较平均帧增益来选择最大扬声器的系统和方法
    • US06125343A
    • 2000-09-26
    • US865399
    • 1997-05-29
    • Guido M. Schuster
    • Guido M. Schuster
    • G10L21/02G06F7/08
    • G10L21/028
    • An improved system for identifying the loudest speech signal in a G.723.1 based audio teleconferencing link is disclosed. The system selects the loudest of several analog audio signals by directly analyzing the encoded G.723.1 bit streams representing those signals, rather than by decoding the encoded speech signal in the G.723.1 bit streams and then re-encoding the signal as a selected output bit stream. The system uses the excitation gain parameters encoded in G.723.1 frames to approximate frame gains for respective bit streams and then estimates a short term speech energy for each bit stream by averaging the approximate frame gains over time. The system then compares the estimated speech energy levels and outputs to each conference participant the signal with the highest estimated speech energy as the next portion of an output signal.
    • 公开了一种用于识别基于G.723.1的音频电话会议链路中最响亮的语音信号的改进系统。 系统通过直接分析代表这些信号的编码的G.723.1比特流而不是通过对G.723.1比特流中的编码语音信号进行解码,然后将该信号重新编码为选择的输出,来选择最大的几个模拟音频信号 位流。 该系统使用G.723.1帧中编码的激励增益参数来近似各个比特流的帧增益,然后通过对随时间推移的近似帧增益进行平均来估计每个比特流的短期语音能量。 然后,系统将估计的语音能级和输出与具有最高估计语音能量的信号作为输出信号的下一部分比较。
    • 9. 发明授权
    • System and method for accessing radio programs using a data network telephone in a network based telecommunication system
    • 用于在基于网络的电信系统中使用数据网络电话访问无线电节目的系统和方法
    • US06914897B1
    • 2005-07-05
    • US09516269
    • 2000-02-29
    • Guido M. SchusterIkhlaq S. SidhuFrederick D. DeanAndrew Bezaitis
    • Guido M. SchusterIkhlaq S. SidhuFrederick D. DeanAndrew Bezaitis
    • H04H1/00H04H20/82H04L12/66H04M1/247H04M1/253H04M7/00
    • H04H20/82H04M1/2473H04M1/2535H04M7/122H04M2250/02
    • A system and method of accessing radio programming from radio stations that communicate radio programming on a data network. The radio programming is accessed as radio or audio signals formatted in radio-over-data packets to a data network telephone. The data network telephone is a telephone that uses voice-over-data communications channels over a data network to make telephone connections with other data network telephones. The data network telephone includes a display, a keypad, a handset and an optional speaker output. The data network telephone advantageously permits simultaneous access to radio programming and communication on a telephone connection. The data network telephone also includes an interface to a portable information device. A radio control application may be used on the portable information device that communicates control information to a radio application on the data network telephone. The PID may be used to set the desired radio station, volume and other settings that may be communicated to the radio application on the data network telephone.
    • 一种从在数据网络上传送无线电节目的无线电站接入无线电节目的系统和方法。 将无线电节目作为以无线电数据包格式化成数据网络电话的无线电或音频信号进行访问。 数据网络电话是使用数据网络上的语音数据通信信道与其他数据网络电话进行电话连接的电话。 数据网络电话包括显示器,键盘,手机和可选的扬声器输出。 数据网络电话有利地允许在电话连接上同时访问无线电节目和通信。 数据网络电话还包括到便携式信息设备的接口。 无线电控制应用可以在向数据网络电话上的无线电应用传送控制信息的便携式信息设备上使用。 PID可以用于设置可以传送到数据网络电话上的无线电应用的期望的无线电台,音量和其他设置。
    • 10. 发明授权
    • Method and system for forward error correction with different frame sizes
    • 具有不同帧大小的前向纠错方法和系统
    • US06785261B1
    • 2004-08-31
    • US09322816
    • 1999-05-28
    • Guido M. SchusterIkhlaq S. SidhuMichael S. BorellaThomas J. Kostas
    • Guido M. SchusterIkhlaq S. SidhuMichael S. BorellaThomas J. Kostas
    • H04L1266
    • H04M7/0072G10L19/005H04L29/06027H04L65/604H04L65/607H04L65/80H04M7/0069
    • A mechanism for recovering data associated with lost packets, suitable for use in a VoIP network. The telecommunications network is preferably a packet switched network having IP telephony gateways serving as interfaces between a telephone device and the IP network. The IP telephony gateway receives a conversation signal from the telephone device, and implements an improved forward error correction method. The method includes generating payload information defined by at least two packet sequences from the same audio information, and transmitting those two packet sequences on the IP network for receipt by a remote network device. The packet sequences are transmitted using RTP with two independent data streams or, alternatively, using a single data stream. The first and second data streams are data packet streams each defining a sequence of data packets. The first data stream is preferably formed using a G.711 vocoder, and the second data stream is preferably formed using a G.723.1 vocoder. The receiver inserts the G.711 packets into a receive buffer, and, in the event that G.711 data is missing or corrupted, the receiver will decode the relevant G.723.1 packets and place it in the buffer in the appropriate location. The buffered data is then used to reproduce the audio information at the receiver.
    • 用于恢复与丢失的数据包相关联的数据的机制,适用于VoIP网络。 电信网络优选地是具有作为电话设备和IP网络之间的接口的IP电话网关的分组交换网络。 IP电话网关从电话装置接收对话信号,实现改进的前向纠错方法。 该方法包括从相同的音频信息生成由至少两个分组序列定义的有效载荷信息,并在IP网络上发送这两个分组序列以供远程网络设备接收。 使用具有两个独立数据流的RTP或者使用单个数据流来发送分组序列。 第一和第二数据流是每个定义数据分组序列的数据分组流。 第一数据流优选地使用G.711声码器形成,并且第二数据流优选地使用G.723.1声码器形成。 接收器将G.711数据包插入接收缓冲区,并且在G.711数据丢失或损坏的情况下,接收器将对相关的G.723.1数据包进行解码并将其放入缓冲区中。 缓冲的数据然后用于在接收机处再现音频信息。