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公开(公告)号：US5790641A
公开(公告)日：1998-08-04
申请号：US666800
申请日：1996-06-19
CPC分类号： H04N1/00312 , H04L47/10 , H04N1/00281 , H04N1/32795 , H04N2201/0015 , H04N2201/0025 , H04N2201/0034
摘要： A system and method for extending the time period at which a response signal can be received before a time-out occurs without altering the length of the time-out period defined in any transmission protocol, ensuring that the two faxes communicate using one of a predefined, e.g., standard, set of transmission protocols, and/or ensuring that the two faxes communicate using one of a predefined set of signal modulation techniques. The present invention receives signals transmitted from a local fax and transmits these signals to a remote fax. Similarly, the present invention receives signals transmitted from the remote fax and transmits these signals to the local fax. The present invention examines the contents of the signals and can modify the signal transmitted to the local fax in order to ensure that the two faxes communicate using a transmission protocol that is supported by the present invention and to ensure that the two faxes communicate using a signal modulation technique that is supported by the present invention.
摘要翻译： 一种用于延长在超时之前可以接收到响应信号的时间段的系统和方法，而不改变在任何传输协议中定义的超时周期的长度，确保两个传真使用预定义的一个进行通信 ，例如标准的传输协议集合和/或确保两个传真使用预定义的一组信号调制技术之一进行通信。 本发明接收从本地传真发送的信号并将这些信号发送到远程传真。 类似地，本发明接收从远程传真发送的信号并将这些信号发送到本地传真。 本发明检查信号的内容，并且可以修改发送到本地传真的信号，以便确保两个传真使用本发明支持的传输协议进行通信，并且确保两个传真使用信号进行通信 本发明支持的调制技术。
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公开(公告)号：US5949861A
公开(公告)日：1999-09-07
申请号：US59635
申请日：1998-04-13
CPC分类号： H04N1/00312 , H04L47/10 , H04N1/00281 , H04N1/32795 , H04N2201/0015 , H04N2201/0025 , H04N2201/0034
摘要： A system and method for extending the time period at which a facsimile response signal can be received before a time-out occurs without altering the underlying transmission protocol. The disclosed system and method determines a response time before which a response signal must be received by a transmitting facsimile station in order to avoid a time-out event. Before the response time arrives, the disclosed system and method generates and transmits a delay signal to the transmitting facsimile station which does not substantively interfere with ongoing communications. Instead, the delay signal causes the transmitting facsimile station to tickle an internal time-out counter which results in an extension of the response signal delay tolerance. Delay signal frames can be transmitted repeatedly to accommodate any desired time-out period.
摘要翻译： 一种用于延长在超时之前可以接收传真响应信号的时间段的系统和方法，而不改变底层的传输协议。 所公开的系统和方法确定响应时间，在此之前响应信号必须由发送传真站接收，以避免超时事件。 在响应时间到达之前，所公开的系统和方法产生并发送延迟信号到发送传真站，该传真机不会实质上干扰正在进行的通信。 相反，延迟信号使得发送传真站发出内部超时计数器，这导致响应信号延迟容限的扩展。 可以重复发送延迟信号帧以适应任何期望的超时周期。
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3.发明授权System and method for reliability transporting aural information across a network 失效通过网络可靠地传送口头信息的系统和方法公开(公告)号：US06298057B1
公开(公告)日：2001-10-02
申请号：US08634927
申请日：1996-04-19
申请人： Kenneth R. Guy , Jaswant R. Jain , Ishwar V. Jasuja , Michael W. Johnson , Albert Juandy , Simon S. Lam , Anthony Y. Lee , David Misunas , Jacques A. Roth
发明人： Kenneth R. Guy , Jaswant R. Jain , Ishwar V. Jasuja , Michael W. Johnson , Albert Juandy , Simon S. Lam , Anthony Y. Lee , David Misunas , Jacques A. Roth
IPC分类号： H04L1228
CPC分类号： H04L12/2854 , H04J3/0632 , H04L1/0002 , H04L1/004 , H04M7/006 , Y02D50/10
摘要： A system and method for transparently transmitting aural signals across a wide area network (WAN). The system of present invention is connected to one or more of a private branch exchange, a key telephone system, a telephone, a facsimile machine, and a modem, for example. In the case of voice transmission, a user places a telephone call using the same procedure that is used when placing a telephone call over a conventional public switched network. The aural signals are translated into a format that is compatible with the local area network (LAN) and the translated signals are transmitted to a router or a switch that connects the LAN to the WAN. The data is transmitted across the WAN to a router or switch coupled to a second LAN. The data is then sent to a destination central site unit or PC which translates the signal into a format that is compatible with the telephone system connected thereto. The present invention provides a voice quality that approaches, equals, or exceeds the voice quality of conventional telephone switched networks. This high voice quality is achieved by utilizing a high quality voice digitization algorithm, by ensuring a low maximum network delay, by dynamically compensating for variations in network delay, and by using a forward error correction technique that can recreate lost or delayed signals in a manner that recreates the signal so the lost signal is typically not detectable by a user.
摘要翻译： 一种用于在广域网（WAN）上透明传输听觉信号的系统和方法。 本发明的系统例如连接到私人交换机，密钥电话系统，电话，传真机和调制解调器中的一个或多个。 在语音传输的情况下，用户使用与在常规公共交换网络上进行电话呼叫时所使用的相同的过程进行电话呼叫。 听觉信号被转换为与局域网（LAN）兼容的格式，并且转换的信号被传送到路由器或将LAN连接到WAN的交换机。 数据通过WAN传输到耦合到第二LAN的路由器或交换机。 然后将数据发送到目的地中心站点单元或PC，其将信号转换为与连接到其的电话系统兼容的格式。 本发明提供接近，等于或超过常规电话交换网络的语音质量的语音质量。 通过使用高质量的语音数字化算法，通过确保低的最大网络延迟，通过动态地补偿网络延迟的变化以及通过使用可以以一种方式重现丢失或延迟的信号的前向纠错技术来实现这种高的语音质量 这会重建信号，使得丢失的信号通常不被用户检测到。
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4.发明授权System and method for transmitting aural information between a computer and telephone equipment 失效用于在计算机和电话设备之间传输听觉信息的系统和方法
公开(公告)号：US5940479A
公开(公告)日：1999-08-17
申请号：US724655
申请日：1996-10-01
申请人： Kenneth R. Guy , Jaswant R. Jain , Ishwar V. Jasuja , Michael W. Johnson , Albert Juandy , Simon S. Lam , Anthony Y. Lee , David Misunas , Jacques A. Roth
发明人： Kenneth R. Guy , Jaswant R. Jain , Ishwar V. Jasuja , Michael W. Johnson , Albert Juandy , Simon S. Lam , Anthony Y. Lee , David Misunas , Jacques A. Roth
CPC分类号： H04L65/605 , H03M13/00 , H04L29/06027 , H04L65/103 , H04L65/104 , H04L65/1069 , H04M7/1205
摘要： A system and method for transmitting aural signals across a wide area network (WAN) from a local phone coupled to a computer, e.g., a PC-phone, to a remote phone coupled to a KTS, PBX, or PSTN, for example. This capability is provided by the gateway unit of the present invention. The system of the present invention is quickly installed in a server or a personal computer coupled to a local area network. The system is connected to one or more of a PSTN, a private branch exchange, a key telephone system, a telephone, a facsimile machine, and a modem. In the case of voice transmission, a gateway unit translates received telephony signal into a format that is compatible with the telephone system or equipment connected thereto. The present invention can provide a voice quality that approaches, equals, or exceeds the voice quality of conventional telephone switched networks. This high voice quality is achieved by utilizing a high quality voice digitization algorithm on both the PC-phone and the gateway, by ensuring a low maximum network delay between the PC-phone and the gateway, by permitting a dynamic compensation for variations in network delay, and by utilizing a forward error correction technique for transmission over WANs that can recreate lost or delayed signals in a manner that recreates the signal so the lost signal is typically not detectable by a user.
摘要翻译： 例如，用于从耦合到诸如PC电话的计算机的本地电话到广域网（WAN）发送听觉信号到例如耦合到KTS，PBX或PSTN的远程电话的系统和方法。 该能力由本发明的网关单元提供。 本发明的系统被快速地安装在耦合到局域网的服务器或个人计算机中。 该系统连接到PSTN，私人交换机，密钥电话系统，电话机，传真机和调制解调器中的一个或多个。 在语音传输的情况下，网关单元将接收到的电话信号转换成与电话系统或与其连接的设备兼容的格式。 本发明可以提供接近，等于或超过常规电话交换网络的语音质量的语音质量。 通过在PC电话和网关上使用高质量的语音数字化算法，通过确保PC电话和网关之间的最小网络延迟，通过允许动态补偿网络延迟的变化来实现这种高的语音质量 并且通过利用前向纠错技术在广域网上进行传输，该技术可以以重建信号的方式再现丢失或延迟的信号，使得丢失的信号通常不被用户检测到。
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5.发明授权Voice/fax digitizing rate negotiation for switched connection in a network environment 失效用于网络环境中交换连接的语音/传真数字化速率协商
公开(公告)号：US5511074A
公开(公告)日：1996-04-23
申请号：US209544
申请日：1994-03-09
申请人： Simon S. Lam , Herbert M. Wildfeuer
发明人： Simon S. Lam , Herbert M. Wildfeuer
CPC分类号： H04N1/33323 , H04J3/247 , H04L5/1446 , H04M11/06 , H04Q11/04 , H04N2201/3335 , H04N2201/33357
摘要： A calling station sends to a called station a call request package including the address of the called station and the information digitizing rate of the calling station. The packet may also include the address of the calling station. The called station selects the lower of the information digitizing rates of the calling and called stations or the information digitizing rates of both stations if such rates are the same. The called station sends to the calling station a call request response package including the selected information digitizing rate. The call request response packet may include the addresses of the calling and called stations. The calling station selects the lower of the information digitizing rates of the calling and called stations or the information digitizing rates of both stations if both rates are the same. The information digitizing rates of the calling and called stations may be the voice digitizing rates of both stations. When the calling station selects the information digitizing rate, the calling and called stations send and receive information packets at such information digitizing rates.
摘要翻译： 呼叫站向被叫站发送包括被叫站的地址和呼叫站的信息数字化速率的呼叫请求包。 分组还可以包括呼叫站的地址。 如果这样的速率是相同的，被叫站选择呼叫和被叫站的信息数字化率较低或两个站的信息数字化速率。 被叫站向呼叫台发送包括所选择的信息数字化速率的呼叫请求响应包。 呼叫请求响应分组可以包括呼叫和被叫站的地址。 如果两个速率相同，则主叫台选择呼叫和被叫站的信息数字化率较低或两个站的信息数字化速率。 呼叫和被叫站的信息数字化速率可以是两个站的语音数字化速率。 当呼叫站选择信息数字化速率时，呼叫和被叫站以这种信息数字化速率发送和接收信息包。
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