公开(公告)号：US06675218B1

公开(公告)日：2004-01-06

申请号：US09134496

申请日：1998-08-14

申请人： Jerry Mahler ,  Michael Borella ,  Guido Schuster ,  Ikhlaq Sidhu

发明人： Jerry Mahler ,  Michael Borella ,  Guido Schuster ,  Ikhlaq Sidhu

IPC分类号： G06F1516

CPC分类号： H04L29/06 ,  H04L69/32

摘要： A system for user-space packet modification, including a set of kernel code and a user-level application programming interface (API). The system facilitates creation of a special socket for passing packets between kernel space and user space. The system in turn facilitates creation and application of a packet filter associated with the socket, in order to trap incoming or outgoing packets being processed in the kernel at a designated point in a protocol stack. Once a packet is trapped, it is moved through the socket into user space, thereby at least temporarily preventing the protocol stack from further processing the packet. In user space, an application may operate on the packet, for instance, modifying aspects of the packet or deleting the packet altogether. The system in turn facilitates injection of a packet from user space into kernel space, and into a designated point in the protocol stack for desired stack processing.

摘要翻译： 用于用户空间分组修改的系统，包括一组内核代码和用户级应用程序编程接口（API）。 该系统有助于创建一个特殊的套接字，用于在内核空间和用户空间之间传递数据包。 该系统又促进了与套接字相关联的分组过滤器的创建和应用，以便在协议栈中的指定点捕获在内核中正在处理的进入或传出的分组。 一旦数据包被捕获，它将通过套接字移动到用户空间中，从而至少暂时防止协议栈进一步处理数据包。 在用户空间中，应用可以对分组进行操作，例如，修改分组的方面或完全删除分组。 该系统又有助于将分组从用户空间注入到内核空间中，并且进入协议栈中的指定点以进行所需的堆栈处理。

公开(公告)号：US06567399B1

公开(公告)日：2003-05-20

申请号：US09305870

申请日：1999-05-05

申请人： Guido Schuster ,  Ikhlaq Sidhu ,  Jerry Mahler ,  Jacek Grabiec

发明人： Guido Schuster ,  Ikhlaq Sidhu ,  Jerry Mahler ,  Jacek Grabiec

IPC分类号： H04J1300

CPC分类号： H04L65/1069 ,  H04L29/06027 ,  H04L65/607 ,  H04M3/007 ,  H04M7/125

摘要： A high-fidelity voice/audio communication system including a high-fidelity SLIC (HSLIC) device that combines traditional BORSCHT functionality with high fidelity sampling and compression techniques. The HSLIC preferably resides on a single plug-in line card contained within a multi-cards chassis. The line card includes an analog interface that connects to a two-wire subscriber line, a high fidelity codec for sampling the analog signal at a high resolution and converting high rate digital signals to an analog signal, a voice processing client running on a microprocessor and associated digital memory. The high fidelity codec preferably has a sample rate of at least twenty thousand samples per second, and no less that 250 quantization levels. The voice processing client preferably includes an Internet Protocol (IP) processing client, Session Initiation Protocol (SIP) client, a Real Time Protocol (RTP) client, and other components of a communication protocol stack for establishing a connection over a packet based network by way of the network interface circuit. The line card establishes a high fidelity audio connection by sending an invite request to a proxy server; receiving an okay signal indicating that the request was received; sending an acknowledge signal; quantizing audio information at a sampling rate greater than twenty thousand samples per second with a resolution of no less than 4096 quantization levels; and, packetizing the quantized data for transmission to a remote device.

摘要翻译： 高保真语音/音频通信系统，包括高传真BORSCHT功能和高保真采样和压缩技术的高保真SLIC（HSLIC）设备。 HSLIC优选地驻留在包含在多卡机箱中的单个插入式线卡上。 线卡包括连接到两线用户线的模拟接口，高保真编解码器，用于以高分辨率对模拟信号进行采样并将高速数字信号转换为模拟信号，在微处理器上运行的语音处理客户机， 相关的数字存储器。 高保真编解码器优选地具有每秒至少二万个样本的采样率，并且不小于250个量化级别。 语音处理客户机优选地包括因特网协议（IP）处理客户端，会话发起协议（SIP）客户端，实时协议（RTP）客户端以及用于通过基于分组的网络建立连接的通信协议栈的其他组件， 网络接口电路的方式。 线卡通过向代理服务器发送邀请请求来建立高保真音频连接; 接收到指示接收到该请求的可信号; 发送确认信号; 以大于每秒二万个样本的采样率量化音频信息，分辨率不低于4096个量化级别; 并且将量化的数据打包以便传输到远程设备。

公开(公告)号：US06487690B1

公开(公告)日：2002-11-26

申请号：US09707567

申请日：2000-11-06

申请人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

发明人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

IPC分类号： H03M1300

CPC分类号： H04L1/008 ,  H04L1/0057

摘要： 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.

摘要翻译： 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。

公开(公告)号：US6151636A

公开(公告)日：2000-11-21

申请号：US18872

申请日：1998-02-05

申请人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

发明人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

IPC分类号： G06F5/00 ,  H03M13/00

CPC分类号： H04L1/004

摘要： A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A data stream representing a media signal at a given level of compression is processed just before the data stream enters the internet. A less compressed data stream representing the same media signal is generated transmitted through the internet. Due to the lower level of compression, the underlying media signal is less sensitive to packet loss in the internet and, as a result, the media signal that arrives at the receiving end will tend to be more continuous and clear.

摘要翻译： 公开了一种通过互联网提高端到端数据或实时媒体传输的速度和质量的方法和装置。 在数据流进入互联网之前处理表示在给定压缩水平下的媒体信号的数据流。 通过互联网发送表示相同媒体信号的较少压缩的数据流。 由于较低的压缩水平，底层媒体信号对互联网中的分组丢失较不敏感，结果，到达接收端的媒体信号往往会更加连续和清晰。

公开(公告)号：US6145109A

公开(公告)日：2000-11-07

申请号：US989616

申请日：1997-12-12

申请人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

发明人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

IPC分类号： H03M13/00

CPC分类号： H03M13/373 ,  H04L1/0041 ,  H04L1/0057 ,  H04L1/0084 ,  H03M13/1515

摘要： 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.

公开(公告)号：US06243846B1

公开(公告)日：2001-06-05

申请号：US09062435

申请日：1998-04-17

申请人： Guido M. Schuster ,  Michael Borella ,  Jerry Mahler ,  Ikhlaq Sidhu

发明人： Guido M. Schuster ,  Michael Borella ,  Jerry Mahler ,  Ikhlaq Sidhu

IPC分类号： H03M1300

CPC分类号： H04L1/004 ,  H03M13/1515 ,  H03M13/373

摘要： 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. For each window of k data packets, the invention generates and transmits at least one cross-wise parity packet taken as an index-shifted function over the k data packets. The invention thereby enables a receiving end to recover from packet loss.

摘要翻译： 一种用于在分组交换网络（诸如因特网）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上简单而强大的前向纠错码方案。 对于k个数据分组的每个窗口，本发明生成并发送在k个数据分组上被用作索引移位函数的至少一个交叉奇偶校验分组。 因此，本发明使得接收端能够从分组丢失中恢复。

公开(公告)号：US06226769B1

公开(公告)日：2001-05-01

申请号：US09243955

申请日：1999-02-03

申请人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

发明人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

IPC分类号： H03M1300

CPC分类号： H04L1/008 ,  H03M13/2703 ,  H03M13/373 ,  H04L1/0041 ,  H04L1/0056

摘要： 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. The invention appends to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload 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.

摘要翻译： 一种用于在分组交换网络（诸如因特网）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上简单而强大的前向纠错码方案。 本发明通过采用先前指定数量的有效载荷分组的异或并定义的单个前向纠错码来附加到一系列有效载荷分组中的每一个。 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。

公开(公告)号：US06170075A

公开(公告)日：2001-01-02

申请号：US08993505

申请日：1997-12-18

申请人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

发明人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

IPC分类号： H03M1300

CPC分类号： H03M13/05 ,  H03M13/09 ,  H03M13/1515 ,  H04L1/08 ,  H04L2001/0097

摘要： A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A media stream being transmitted to the internet is channel coded at the edge of the internet in order to free upstream bit rate for use in source coding the media. The channel coded media stream may then be decoded at a remote edge of the internet to recover lost packets.

摘要翻译： 公开了一种通过互联网提高端到端数据或实时媒体传输的速度和质量的方法和装置。 正在传输到因特网的媒体流在因特网的边缘进行信道编码，以便释放用于媒体源编码的上行比特率。 然后可以在互联网的边缘边解码信道编码的媒体流，以恢复丢失的分组。

公开(公告)号：US5870412A

公开(公告)日：1999-02-09

申请号：US989483

申请日：1997-12-12

申请人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

发明人： Guido M. Schuster ,  Jerry Mahler ,  Ikhlaq Sidhu ,  Michael Borella

IPC分类号： G06F11/00 ,  H03M13/00 ,  H03M13/27

CPC分类号： H04L1/008 ,  H03M13/2703 ,  H03M13/373 ,  H04L1/0041 ,  H04L1/0056

摘要： 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. The invention appends to each of a series of payload packets a single forward error correction code that is defined by taking the XOR sum of a preceding specified number of payload 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.

摘要翻译： 一种用于在分组交换网络（诸如因特网）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上简单而强大的前向纠错码方案。 本发明通过采用先前指定数量的有效载荷分组的异或并定义的单个前向纠错码来附加到一系列有效载荷分组中的每一个。 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。

公开(公告)号：US06570606B1

公开(公告)日：2003-05-27

申请号：US09087155

申请日：1998-05-29

申请人： Ikhlaq Sidhu ,  Guido M. Schuster ,  Jerry Mahler ,  Thomas Kostas

发明人： Ikhlaq Sidhu ,  Guido M. Schuster ,  Jerry Mahler ,  Thomas Kostas

IPC分类号： H04N714

CPC分类号： H04N7/141

摘要： A method and apparatus for controlling the transmission of real-time media signals over a data network based on a triggering event associated with a participating teleconference station. This triggering event may be the presence or absence of motion, the presence or absence of sound, or any of a variety of other events that preferably indicate the presence or absence of a person at the station. When no person is present at the station and/or when an appropriate triggering event occurs, remote teleconference participants will not transmit media signals over the network to the station, thereby conserving valuable network bandwidth and simulating a physically common meeting between people. In turn, when a person is present at the station and/or when another appropriate triggering event occurs, remote teleconference participants may start transmitting media signals over the network to the station. Additionally, a station may control its own transmission of media signals to remote stations in response to local triggering events, thus further conserving network resources.

摘要翻译： 一种用于基于与参与电话会议站相关联的触发事件来控制数据网络上的实时媒体信号的传输的方法和装置。 该触发事件可以是运动的存在或不存在，声音的存在或不存在，或者优选地指示在车站处存在或不存在人的各种其他事件中的任何一种。 当没有人在车站和/或发生适当的触发事件时，远程电话会议参与者将不会通过网络将媒体信号发送到车站，从而节省宝贵的网络带宽并模拟人们之间的物理上常见的会议。 反过来，当一个人出现在车站和/或当另一个适当的触发事件发生时，远程电话会议参与者可以开始通过网络向车站发送媒体信号。 此外，站可以响应于本地触发事件来控制其自己的传输到远程站的媒体信号，从而进一步节省网络资源。