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公开(公告)号：US06351524B1
公开(公告)日：2002-02-26
申请号：US09234187
申请日：1999-01-20
IPC分类号： H04M164
CPC分类号： H04L12/2874 , H04L12/2856 , H04L51/24 , H04M3/537 , H04M15/00 , H04M15/8083 , H04M2203/4536 , H04M2215/0184 , H04M2215/22
摘要： A toll saving method and apparatus for a remote access system is provided. A first communication device receives a connection request from a second communication device. The first communication device then decodes caller identification information from the connection request to determine a user identity, without completing the connection request. The first communication device determines whether an event associated with the user identity has occurred. If the event associated with the user identity has not occurred, the connection request is not completed. In a preferred embodiment, the toll saving method and apparatus allows an email user to call an ISP to determine whether unaccessed email is present in the user's account. If no email is present in the user's account, the call is never answered and no toll charges are incurred.
摘要翻译： 提供了一种用于远程访问系统的长途保存方法和装置。 第一通信设备从第二通信设备接收连接请求。 第一通信设备然后从连接请求中解码呼叫者识别信息以确定用户身份，而不完成连接请求。 第一通信设备确定是否发生了与用户身份相关联的事件。 如果与用户身份相关联的事件未发生，则连接请求未完成。 在优选实施例中，费用保存方法和装置允许电子邮件用户呼叫ISP以确定用户帐户中是否存在未处理的电子邮件。 如果用户帐户中没有电子邮件，则该呼叫永远不会被接收，并且不会产生任何费用。
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2.发明授权公开(公告)号：US06512761B1
公开(公告)日：2003-01-28
申请号：US09241941
申请日：1999-02-02
IPC分类号： H04L1256
CPC分类号： H04L65/1043 , H04L12/14 , H04L12/1403 , H04L12/1414 , H04L12/1485 , H04L12/1496 , H04L29/06 , H04L29/06027 , H04L41/5029 , H04L47/10 , H04L47/2416 , H04L47/283 , H04L47/29 , H04L65/80
摘要： A system for adjusting or establishing transmission fees based on delay and/or jitter. A network element may evaluate transmission delay and/or jitter over a given transmission system, which may be or is in effect for a particular real-time media transmission. A determination may then be made whether the delay and/or jitter exceeds a predetermined threshold, which may suggest that the transmission quality would be lower than desired. If so, then the fees that a billing entity would charge for the transmission may be decreased in view to account for the lesser quality of service. Conversely, where the delay and/or jitter is lower than a predetermined threshold, then the billing entity may increase its fees accordingly. Other fee schedules based on delay may be established as well.
摘要翻译： 基于延迟和/或抖动来调整或建立传输费用的系统。 网络元件可以评估给定传输系统的传输延迟和/或抖动，其可以是或者对于特定的实时媒体传输有效。 然后可以确定延迟和/或抖动是否超过预定阈值，这可能表明传输质量将低于期望值。 如果是这样，则为了考虑更低的服务质量，计费实体将为传输收费的费用可能会减少。 相反，在延迟和/或抖动低于预定阈值的情况下，计费实体可以相应地增加其费用。 也可以建立基于延迟的其他费用表。
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公开(公告)号：US06954454B1
公开(公告)日：2005-10-11
申请号：US09303514
申请日：1999-05-03
申请人： Guido M. Schuster , Ikhlaq S. Sidhu , Jerry J. Mahler , David Grabelsky , Jacek A. Grabiec , Thomas J. Kostas , Michael S. Borella
发明人： Guido M. Schuster , Ikhlaq S. Sidhu , Jerry J. Mahler , David Grabelsky , Jacek A. Grabiec , Thomas J. Kostas , Michael S. Borella
CPC分类号： H04L12/56
摘要： A telephony system and method having a switch for analog voice and data signals that is connected to a first network, and a router for routing Internet Protocol packets that is connected to a second network using Internet Protocol addressing. The telephony system and method also includes a telephony gateway that is connected to both the switch and the router for converting analog voice signals into Internet Protocol packets and for converting Internet Protocol packets into analog voice signals, the telephony gateway being connected, and a remote access server that is connected to both the switch and the router for converting analog data signals into Internet Protocol packets and for converting Internet Protocol packets into analog data signals. The switch may have a switch matrix capable of being connected to the Public Switched Telephone Network, a line rack with a plurality of line cards connected to the switch matrix, and a trunk rack with a plurality of trunk cards connected to the switch matrix. The switch matrix may also be connected to the telephony gateway and the remote access server.
摘要翻译： 具有连接到第一网络的模拟语音和数据信号的交换机的电话系统和方法，以及用于使用因特网协议寻址连接到第二网络的因特网协议分组的路由器。 电话系统和方法还包括连接到交换机和路由器的电话网关，用于将模拟语音信号转换为因特网协议分组，并将互联网协议分组转换为模拟语音信号，连接的电话网关和远程访问 连接到交换机和路由器的服务器，用于将模拟数据信号转换成互联网协议包，并将互联网协议包转换为模拟数据信号。 开关可以具有能够连接到公共交换电话网络的开关矩阵，具有连接到开关矩阵的多个线路卡的线架，以及具有连接到开关矩阵的多个干线卡的中继架。 交换矩阵也可以连接到电话网关和远程访问服务器。
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4.发明授权公开(公告)号：US06360271B1
公开(公告)日：2002-03-19
申请号：US09241689
申请日：1999-02-02
IPC分类号： G06F1516
CPC分类号： H04L65/604 , H04J3/0632 , H04J3/0638 , H04J3/0644 , H04J3/0682 , H04L29/06027 , H04L49/206 , H04L49/25 , H04L49/30 , H04L65/80 , H04L2012/5674
摘要： A system for dynamically jitter buffering a sequence of packets based on substantially synchronized time signals maintained at the transmitting and receiving ends of a communication system. The time signals may be synchronized, for instance, by equipping both the transmitting and receiving ends with global positioning system receivers. The transmitting end may mark each outgoing packet with a sender-time based on the time signal at the transmitting end. Dynamic jitter buffering may then be provided by scheduling delayed play-out of each packet. For instance, the jitter buffer at the receiving end may be configured to delay play-out of each packet until the time signal at the receiving end indicates a time that is substantially a predetermined end-to-end delay period after the sender-time for the packet.
摘要翻译： 一种用于基于在通信系统的发送和接收端保持的基本上同步的时间信号来动态地抖动缓冲分组序列的系统。 时间信号可以例如通过将发送和接收端装备到全球定位系统接收器来同步。 发送端可以基于发送端的时间信号以发送方时间标记每个输出分组。 然后可以通过调度每个分组的延迟播出来提供动态抖动缓冲。 例如，接收端的抖动缓冲器可以被配置为延迟每个分组的播出，直到接收端的时间信号指示在发送者时间之后基本上是预定的端到端延迟周期的时间 包。
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5.发明授权Methods for determining sendable information content based on a determined network latency 有权基于确定的网络延迟确定可发送信息内容的方法公开(公告)号：US06182125B2
公开(公告)日：2001-01-30
申请号：US09170437
申请日：1998-10-13
IPC分类号： G06F1516
CPC分类号： G06F17/30905
摘要： A method for improving perception of electronic content from a computer network such as the Internet or an intranet. Network latencies and the type of electronic content such as text, graphical images, animation, voice, video and other electronic content interact to influence user perception of the quality of information provided. As network latency increases and becomes more variable, users typically become less satisfied. The method dynamically adjusts the amount of electronic content presented to user based on a determined network latency. The amount of electronic content is also adjusted progressively and underlying transport protocol such as Transmission Control Protocol (“TCP”) and User Datagram Protocol (“UDP”) are adaptively adjusted based on the type of electronic content requested (e.g., TCP for text, UDP for graphical images, etc.). The method may improve user perception of requested original electronic content by dynamically sending an amount of original electronic content based on a determined network latency. Improved user perception of original electronic content may help attract and retain, students, customers, contributors, etc. to an organization's electronic content site on a computer network (e.g., a home page on the Internet or an intranet).
摘要翻译： 一种用于改善来自诸如因特网或内部网的计算机网络的电子内容的感知的方法。 网络延迟和诸如文本，图形图像，动画，语音，视频等电子内容的电子内容的类型交互以影响用户对提供的信息质量的感知。 随着网络延迟增加并变得更加变数，用户通常会变得不那么满意。 该方法基于确定的网络延迟动态地调整呈现给用户的电子内容的量。 电子内容的数量也被逐渐调整，并且基于所请求的电子内容的类型（例如，用于文本的TCP）来自适应地调整诸如传输控制协议（“TCP”）和用户数据报协议（“UDP” UDP用于图形图像等）。 该方法可以基于确定的网络延迟动态地发送一定量的原始电子内容来改善用户对所请求的原始电子内容的感知。 改善用户对原始电子内容的看法可能有助于吸引和留住计算机网络（例如，互联网或内部网上的主页）上的组织的电子内容站点，学生，客户，贡献者等。
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公开(公告)号：US06771674B1
公开(公告)日：2004-08-03
申请号：US09221752
申请日：1998-12-28
IPC分类号： H04J302
CPC分类号： H04L65/1026 , H04L1/0041 , H04L1/0057 , H04L1/0083 , H04L29/06027 , H04L65/1036 , H04L65/605 , H04L65/607 , H04L65/80 , H04L69/14
摘要： A mechanism for forward error correction (FEC) coding, suitable for use where multiple payload streams are simultaneously transmitted from end-to-end. Instead of deriving parity information based on payload information carried within a given stream, the invention involves FEC encoding across multiple parallel streams and thereby deriving parallel parity information. The parallel parity information may then be transmitted to the receiving end in parallel with the underlying payload information. Beneficially, the invention can substantially reduce the time it takes for the transmitting end to derive parity information or for the receiving end to receive the information necessary to recover from data loss. The invention is especially suitable for use in IP telephony and particularly for implementation in an IP telephony gateway.
摘要翻译： 用于前向纠错（FEC）编码的机制，适用于从端到端同时发送多个有效负载流的情况。 本发明不是基于给定流中携带的有效载荷信息导出奇偶校验信息，而是涉及跨多个并行流的FEC编码，从而导出并行奇偶校验信息。 然后可以将平行奇偶校验信息与底层有效载荷信息并行发送到接收端。 有利地，本发明可以显着地减少发送端导出奇偶校验信息所需的时间，或者为了接收端接收从数据丢失恢复所必需的信息。 本发明特别适用于IP电话，特别适用于IP电话网关中的实现。
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7.发明授权Method and apparatus for measurement-based conformance testing of service level agreements in networks 有权网络中服务级别协议的基于测量的一致性测试的方法和装置公开(公告)号：US06363053B1
公开(公告)日：2002-03-26
申请号：US09246606
申请日：1999-02-08
IPC分类号： G01R3108
CPC分类号： H04L41/5003 , H04L41/5009 , H04L43/50 , H04L47/24
摘要： A method and apparatus for measurement-based conformance testing of service level agreements in networks. The method includes first collecting quality of service information from network traffic over a plurality of network nodes. Then, the collected quality of service information is compared to a plurality of specified quality of service levels. A plurality of possible virtual quality of service pathways through a plurality of network nodes is provided, based on the compared quality of service information. One embodiment of the method includes the additional step of creating a virtual connection using the compared quality of service information. In another embodiment of the method, the step of collecting quality of service information from network traffic over a plurality of network nodes includes first transmitting test traffic from a source to a destination over a plurality of network nodes. The transmitted test traffic is then received at the destination, and quality of service information is identified by comparing characteristics of the test traffic transmitted by the source to characteristics of the test traffic received by the destination.
摘要翻译： 一种用于网络中服务级别协议的基于测量的一致性测试的方法和装置。 该方法包括首先通过多个网络节点从网络流量收集服务质量信息。 然后，将所收集的服务质量信息与多个指定的服务质量水平进行比较。 基于所比较的服务质量信息，提供通过多个网络节点的多个可能的虚拟服务质量路径。 该方法的一个实施例包括使用所比较的服务质量信息创建虚拟连接的附加步骤。 在该方法的另一实施例中，从多个网络节点上的网络业务收集服务质量信息的步骤包括首先通过多个网络节点从源向目的地发送测试业务。 然后在目的地接收所发送的测试业务，并且通过将源发送的测试业务的特征与目的地接收到的测试业务的特性进行比较来识别服务质量信息。
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8.发明授权Profile based method for packet header compression in a point to point link 有权基于配置文件的方法，用于在点对点链接中进行数据包头压缩公开(公告)号：US06542504B1
公开(公告)日：2003-04-01
申请号：US09322845
申请日：1999-05-28
IPC分类号： H04L1228
摘要： A method is shown for compression of packet header information of packets transmitted on a point to point link. First and second endpoints of the point to point link negotiate a profile for packet header information for packets transmitted from the first endpoint to the second endpoint on the point to point link. The profile includes a predetermined default value for a predetermined header field of the packet header information. A packet sent from the first endpoint to the second endpoint over the point to point link includes a profile identifier for the profile and excludes the predetermined header field. The second endpoint uses the profile identifier to access the profile. The second endpoint then uses the predetermined default value for the predetermined header field from the profile to decode the packet.
摘要翻译： 示出了用于压缩在点对点链路上发送的分组的分组报头信息的方法。 点对点链路的第一和第二端点协商用于在点对点链路上从第一端点发送到第二端点的分组的分组报头信息的简档。 该简档包括用于分组报头信息的预定报头字段的预定默认值。 通过点对点链路从第一端点发送到第二端点的分组包括用于简档的简档标识符并排除预定的报头字段。 第二个端点使用配置文件标识符来访问配置文件。 然后，第二端点使用来自简档的预定标头字段的预定默认值来解码分组。
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公开(公告)号：US06822957B1
公开(公告)日：2004-11-23
申请号：US09707708
申请日：2000-11-07
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。
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公开(公告)号：US06731642B1
公开(公告)日：2004-05-04
申请号：US09303832
申请日：1999-05-03
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。
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