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31. 发明授权

US06487603B1 Method and apparatus for real time communication over switched networks 失效
标题翻译：用于交换网络实时通信的方法和装置
公开(公告)号：US06487603B1
公开(公告)日：2002-11-26
申请号：US09692499
申请日：2000-10-19
申请人： Guido M. Schuster , Ikhlaq S. Sidhu
发明人： Guido M. Schuster , Ikhlaq S. Sidhu
IPC分类号： G06F1300
CPC分类号： H04L65/604 , H04J3/0682 , H04L12/6418 , H04L65/80 , H04L2012/5616 , H04L2012/5647 , H04L2012/5649 , H04L2012/5671 , H04L2012/5681 , H04L2012/6429 , H04L2012/6472 , H04L2012/6475 , H04L2012/6481 , H04L2012/6483 , H04L2012/6489 , H04Q11/0478
摘要： A method and apparatus for communicating a real time media input over a network. The apparatus encodes the input into data packets having a number of frames ordered according to a first variable. A receiving device unpacks and buffers the unpacked data packets for playout according to a second variable. The receiving device generates utility parameters for evaluating a dynamic characteristic of the network that transports the data packets. The receiving device selects a preferred utility parameter and adjusts the first and second variable according to the selected utility parameter. The method includes encoding an analog input into data packets that are transported to a receiving device. The method also includes unpacking the data packets, buffering the unpacked data packets according to a second variable, and generating at least two utility parameters that represent a dynamic characteristic of a network. The method also includes selecting a preferred utility parameter and adjusting the first and the second variables according to the selected preferred utility parameter.
摘要翻译：一种用于在网络上传送实时媒体输入的方法和装置。该装置将输入编码为具有根据第一变量排序的多个帧的数据分组。接收设备根据第二变量解包和缓冲未打包的数据包进行播出。接收设备产生用于评估传输数据分组的网络的动态特性的实用参数。接收装置选择优选的实用参数，并根据选择的效用参数调整第一和第二变量。该方法包括将模拟输入编码到被传输到接收设备的数据分组中。该方法还包括解包数据分组，根据第二变量缓冲未打包的数据分组，以及生成表示网络的动态特性的至少两个效用参数。该方法还包括选择优选实用参数并根据所选择的优选实用参数来调整第一和第二变量。

32. 发明授权

US06822957B1 Distributed network address translation for a network telephony system 失效
标题翻译：网络电话系统的分布式网络地址转换
公开(公告)号：US06822957B1
公开(公告)日：2004-11-23
申请号：US09707708
申请日：2000-11-07
申请人： Guido M. Schuster , Michael S. Borella , David A. Grabelsky , Ikhlaq S. Sidhu
发明人： Guido M. Schuster , Michael S. Borella , David A. Grabelsky , Ikhlaq S. Sidhu
IPC分类号： H04L1228
CPC分类号： H04L29/12009 , H04L29/12377 , H04L29/12415 , H04L29/125 , H04L29/12528 , H04L61/2517 , H04L61/2532 , H04L61/2564 , H04L61/2575 , H04L63/0823
摘要： System and method for distributed network address translation in a network telephony system. A first network phone with a first protocol, requests at least one locally unique port from a first network device. The first network phone and the first network device are located on a first network. The first network phone receives, with the first protocol, the at least one locally unique port from the first network device. At least one default or ephemeral port on the first network phone is replaced with the at least one locally unique port. A combination network address is created for the first network phone with the at least one locally unique port and a common external network address, thereby identifying the first network phone for communications with a second network device located on a second network. The second network device may, for example, be a second network phone. In a preferred embodiment, the first protocol is a Port Allocation Protocol, such as the Realm Specific Internet Protocol.
摘要翻译：网络电话系统中分布式网络地址转换的系统和方法。具有第一协议的第一网络电话请求从第一网络设备至少一个本地唯一的端口。第一网络电话和第一网络设备位于第一网络上。第一网络电话利用第一协议接收来自第一网络设备的至少一个本地唯一端口。第一个网络电话上至少有一个默认或临时端口被替换为至少一个本地唯一端口。为具有至少一个本地唯一端口和公共外部网络地址的第一网络电话创建组合网络地址，从而识别用于与位于第二网络上的第二网络设备进行通信的第一网络电话。第二网络设备可以例如是第二网络电话。在优选实施例中，第一协议是诸如领域特定因特网协议之类的端口分配协议。

33. 发明授权

US06744759B1 System and method for providing user-configured telephone service in a data network telephony system 有权
标题翻译：用于在数据网络电话系统中提供用户配置的电话服务的系统和方法
公开(公告)号：US06744759B1
公开(公告)日：2004-06-01
申请号：US09405283
申请日：1999-09-27
申请人： Ikhlaq S. Sidhu , Guido M. Schuster , Frederick D. Dean , Andrew Bezaitis
发明人： Ikhlaq S. Sidhu , Guido M. Schuster , Frederick D. Dean , Andrew Bezaitis
IPC分类号： H04L1256
CPC分类号： H04Q3/0033 , H04M3/42 , H04M3/42161 , H04M3/42314 , H04M7/0036 , H04M7/12 , H04M7/1245 , H04M2207/20
摘要： A system and method for providing user-configured telephone service to a user of a data network telephone. The user connects a data network telephone to the data network. The data network telephone registers with a telephone connection server to have basic calling service. The user accesses a service provider server to enter feature selections. The service provider server may use a web page to query the user for feature selections The service provider server uses the user's selections to update the user's account and to activate the selected features.
摘要翻译：一种用于向数据网络电话用户提供用户配置的电话服务的系统和方法。用户将数据网络电话连接到数据网络。数据网络电话与电话连接服务器注册以具有基本呼叫服务。用户访问服务提供商服务器以输入功能选择。服务提供商服务器可以使用网页来查询用户的特征选择服务提供商服务器使用用户的选择来更新用户的帐户并激活所选择的特征。

34. 发明授权

US06731642B1 Internet telephony using network address translation 有权
标题翻译：网络电话使用网络地址转换
公开(公告)号：US06731642B1
公开(公告)日：2004-05-04
申请号：US09303832
申请日：1999-05-03
申请人： Michael S. Borella , Nurettin B. Beser , Ikhlaq S. Sidhu , Guido M. Schuster
发明人： Michael S. Borella , Nurettin B. Beser , Ikhlaq S. Sidhu , Guido M. Schuster
IPC分类号： H04L1228
CPC分类号： H04L61/2514 , H04L29/12009 , H04L29/12367 , H04L29/125 , H04L29/12528 , H04L29/12556 , H04L61/2564 , H04L61/2575 , H04L61/2585 , H04M7/1245
摘要： A system and method for Internet telephony between a caller station and a callee station are described. The caller station is connected to a first edge network via a first telephony interface, and the callee station is connected to a second edge network via a second telephony interface. An intermediate network is connected to the first edge network via a first router and is connected to the second edge network via a second router. The callee station is associated with a callee station number. The first router initiates the call in response to a setup message that includes the callee station number. A first gatekeeper, controlling the first router, and a second gatekeeper, controlling the second router, together mediate the process of setting up the call. A back end server, in communication with the first and second gatekeepers, stores the addresses and station numbers needed to set up the call. During the call, the first router performs network address translation to transmit signals between the first edge network and the Internet, and the second router performs network address translation to transmit signals between the second edge network and the Internet.
摘要翻译：描述了呼叫者站和被叫站之间的因特网电话的系统和方法。呼叫站经由第一电话接口连接到第一边缘网络，被叫站经由第二电话接口连接到第二边缘网络。中间网络经由第一路由器连接到第一边缘网络，并且经由第二路由器连接到第二边缘网络。被叫站与被叫站号相关联。响应于包括被叫站号码的建立消息，第一路由器发起呼叫。控制第一路由器的第一个看门人，以及控制第二个路由器的第二个看门人，一起调解设置呼叫的过程。与第一和第二看门人通信的后端服务器存储建立呼叫所需的地址和站号。在呼叫期间，第一路由器执行网络地址转换以在第一边缘网络和因特网之间传输信号，并且第二路由器执行网络地址转换以在第二边缘网络和因特网之间传送信号。

35. 发明授权

US06681252B1 System and method for interconnecting portable information devices through a network based telecommunication system 有权
标题翻译：通过基于网络的电信系统互连便携式信息设备的系统和方法
公开(公告)号：US06681252B1
公开(公告)日：2004-01-20
申请号：US09406152
申请日：1999-09-27
申请人： Guido M. Schuster , Ikhlaq S. Sidhu , Frederick D. Dean , Ronnen Belkind
发明人： Guido M. Schuster , Ikhlaq S. Sidhu , Frederick D. Dean , Ronnen Belkind
IPC分类号： G06F1300
CPC分类号： H04L65/1026 , H04L65/1006 , H04M7/1205
摘要： A personal information device (PID) is coupled to an IP Telephony phone in order to provide end-to-end connectivity to another PID through a network. The architecture disclosed includes a pair of internet-enabled phones that are able to establish a call session using a Session Initiation Protocol (SIP) and a Session Description Protocol (SDP). Each phone is also provided with an interface configured to communicate with a PID. Each PID is registered to a corresponding internet-enabled phone using each PID user's SIP URL. The user of a first PID connected to a first phone requests a call to a SIP URL corresponding to the user of the second PID that is connected to a second phone. The SIP URL for the user of the second PID is resolved to the network address of the second phone and connection is established between the first and second phones. The connection includes a media stream for transferring data between each of the PIDs. A data object transmitted by the first PID through its interface with the first phone is transmitted to the second phone through the media stream of the connection between the first and second phones. The data object received by the second phone is transmitted to the second PID through the interface between the second phone and the second PID.
摘要翻译：个人信息设备（PID）被耦合到IP电话电话，以便通过网络提供到另一个PID的端到端连接。所公开的架构包括能够使用会话发起协议（SIP）和会话描述协议（SDP）建立呼叫会话的一对因特网的电话。每个电话还具有配置成与PID进行通信的接口。每个PID都使用每个PID用户的SIP URL注册到相应的启用互联网的手机。连接到第一电话的第一PID的用户请求与连接到第二电话的第二PID的用户对应的SIP URL的呼叫。用于第二PID的用户的SIP URL被解析为第二电话的网络地址，并且在第一和第二电话之间建立连接。该连接包括用于在每个PID之间传送数据的媒体流。由第一PID通过其与第一电话的接口发送的数据对象通过第一和第二电话之间的连接的媒体流被发送到第二电话。由第二电话接收的数据对象通过第二电话和第二PID之间的接口发送到第二PID。

36. 发明授权

US06434606B1 System for real time communication buffer management 有权
标题翻译：系统实时通讯缓冲管理
公开(公告)号：US06434606B1
公开(公告)日：2002-08-13
申请号：US09322561
申请日：1999-05-28
申请人： Michael S. Borella , Ikhlaq S. Sidhu , Guido M. Schuster , Jerry J. Mahler , Thomas J. Kostas
发明人： Michael S. Borella , Ikhlaq S. Sidhu , Guido M. Schuster , Jerry J. Mahler , Thomas J. Kostas
IPC分类号： G06F1300
CPC分类号： H04L49/557 , H04J3/0632 , H04J3/0638 , H04J3/0682 , H04L12/6418 , H04L29/06027 , H04L49/206 , H04L49/253 , H04L49/30 , H04L65/604 , H04L65/80 , H04L2012/5616 , H04L2012/5647 , H04L2012/5649 , H04L2012/5671 , H04L2012/5674 , H04L2012/5681 , H04L2012/6429 , H04L2012/6472 , H04L2012/6475 , H04L2012/6481 , H04L2012/6483 , H04L2012/6489 , H04Q11/0478
摘要： A system and method for receiving a transported stream of data packets includes a buffer management device for receiving the data packets, unpacking the data packets, and forwarding a stream of data frames. The system and method further includes a first jitter buffer for receiving the data frames from the buffer management device and buffering the data frames, and a second jitter buffer for receiving the data frames from the buffer management device and buffering the data frames. In addition, the system and method includes a computationally-desirable jitter buffer selected from the first jitter buffer or the second jitter buffer by comparing a first jitter buffer quality and a second jitter buffer quality. The system and method also includes a decoder for receiving buffered data frames from the computationally-desirable jitter buffer.
摘要翻译：用于接收数据分组的传输流的系统和方法包括用于接收数据分组，解包数据分组和转发数据帧流的缓冲器管理装置。该系统和方法还包括第一抖动缓冲器，用于从缓冲器管理装置接收数据帧并缓冲数据帧;以及第二抖动缓冲器，用于从缓冲器管理装置接收数据帧并缓冲数据帧。此外，该系统和方法包括通过比较第一抖动缓冲器质量和第二抖动缓冲器质量从第一抖动缓冲器或第二抖动缓冲器中选择的计算上期望的抖动缓冲器。该系统和方法还包括用于从计算上期望的抖动缓冲器接收缓冲数据帧的解码器。

37. 发明授权

US06175871B1 Method and apparatus for real time communication over packet networks 失效
标题翻译：用于通过分组网络实时通信的方法和装置
公开(公告)号：US06175871B1
公开(公告)日：2001-01-16
申请号：US08942446
申请日：1997-10-01
申请人： Guido M. Schuster , Ikhlaq S. Sidhu
发明人： Guido M. Schuster , Ikhlaq S. Sidhu
IPC分类号： G06F1300
CPC分类号： H04L65/604 , H04J3/0682 , H04L12/6418 , H04L65/80 , H04L2012/5616 , H04L2012/5647 , H04L2012/5649 , H04L2012/5671 , H04L2012/5681 , H04L2012/6429 , H04L2012/6472 , H04L2012/6475 , H04L2012/6481 , H04L2012/6483 , H04L2012/6489 , H04Q11/0478
摘要： A method and apparatus for communicating a real time media input over a network. The apparatus encodes the input into data packets having a number of frames ordered according to a first variable. A receiving device unpacks and buffers the unpacked data packets for playout according to a second variable. The receiving device generates utility parameters for evaluating a dynamic characteristic of the network that transports the data packets. The receiving device selects a preferred utility parameter and adjusts the first and second variable according to the selected utility parameter. The method includes encoding an analog input into data packets that are transported to a receiving device. The method also includes unpacking the data packets, buffering the unpacked data packets according to a second variable, and generating at least two utility parameters that represent a dynamic characteristic of a network. The method also includes selecting a preferred utility parameter and adjusting the first and the second variables according to the selected preferred utility parameter.
摘要翻译：一种用于在网络上传送实时媒体输入的方法和装置。该装置将输入编码为具有根据第一变量排序的多个帧的数据分组。接收设备根据第二变量解包和缓冲未打包的数据包进行播出。接收设备产生用于评估传输数据分组的网络的动态特性的实用参数。接收装置选择优选的实用参数，并根据选择的效用参数调整第一和第二变量。该方法包括将模拟输入编码到被传输到接收设备的数据分组中。该方法还包括解包数据分组，根据第二变量缓冲未打包的数据分组，以及生成表示网络的动态特性的至少两个效用参数。该方法还包括选择优选实用参数并根据所选择的优选实用参数来调整第一和第二变量。

38. 发明授权

US06487690B1 Forward error correction system for packet based real time media 失效
标题翻译：用于基于分组的实时媒体的前向纠错系统
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。

39. 发明授权

US6151636A Data and media communication through a lossy channel using signal conversion 失效
标题翻译：通过使用信号转换的有损通道进行数据和媒体通信
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。

40. 发明授权

US06125343A System and method for selecting a loudest speaker by comparing average frame gains 失效
标题翻译：通过比较平均帧增益来选择最大扬声器的系统和方法
公开(公告)号：US06125343A
公开(公告)日：2000-09-26
申请号：US865399
申请日：1997-05-29
申请人： Guido M. Schuster
发明人： Guido M. Schuster
IPC分类号： G10L21/02 , G06F7/08
CPC分类号： G10L21/028
摘要： 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.
摘要翻译：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。

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