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    • 1. 发明授权
    • Conferencing architecture employing media servers and enhanced session initiation protocol
    • 使用媒体服务器和增强会话发起协议的会议架构
    • US07590692B2
    • 2009-09-15
    • US10191788
    • 2002-07-09
    • John Jeffrey Van DykeAndy Spitzer
    • John Jeffrey Van DykeAndy Spitzer
    • G06F15/16
    • H04L65/1006H04L29/06H04L29/06027H04L65/1096H04L65/608H04L67/02
    • A conferencing system that can access advanced conferencing features while following essentially the same call flow as conventional conferencing systems. The conferencing system includes a computer network, and at least one conferencing application server, at least one media server, and at least one user agent connected to the network. The conferencing application server establishes and manages multimedia conferences by engaging in Session Initiation Protocol (SIP) signaling with the user agents and the media server. Once the conference is established, the media server generates multimedia data such as audio data and conveys the data to the conference participants. In order to access advanced conferencing features, the conferencing system employs an enhanced SIP signaling technique including a conferencing Application Programming Interface (API) implemented by incorporating Extensible Mark-up Language (XML) messages in the bodies of respective SIP request/response messages. The XML messages are incorporated in the SIP request/response message bodies to convey conference specific commands and/or parameters that cannot be easily described via the Session Description Protocol (SDP).
    • 会议系统可以访问高级会议功能,同时遵循与传统会议系统基本相同的呼叫流程。 该会议系统包括计算机网络,以及至少一个会议应用服务器,至少一个媒体服务器以及连接到该网络的至少一个用户代理。 会议应用服务器通过与用户代理和媒体服务器进行会话发起协议(SIP)信令来建立和管理多媒体会议。 一旦会议建立,媒体服务器就产生诸如音频数据的多媒体数据,并将数据传送给会议参与者。 为了访问高级会议功能,会议系统采用增强型SIP信令技术,其包括通过将可扩展标记语言(XML)消息合并在相应的SIP请求/响应消息的主体中而实现的会议应用编程接口(API)。 XML消息被并入SIP请求/响应消息体中以传送会议特定命令和/或不能通过会话描述协议(SDP)容易地描述的参数。
    • 2. 发明授权
    • Jitter buffer management
    • 抖动缓冲管理
    • US07499472B2
    • 2009-03-03
    • US11076815
    • 2005-03-10
    • Andy Spitzer
    • Andy Spitzer
    • H04J3/07H04J3/06
    • H04L47/28H04L12/66H04L29/06027H04L47/2416H04L47/56H04L49/90H04L65/608H04L65/80
    • A sample jitter buffer manager more or less aggressively conserves (rations) or discards data in a jitter buffer, based on the fluctuating amount of data in the jitter buffer. The jitter buffer manager counts, provides, discards and/or otherwise manages individual sample data units, rather than entire packets. Normally, enough data is removed from the jitter buffer to fill a data packet for a receiver. However, if the amount of data in the jitter buffer is low, less data is removed from the jitter buffer and placed into the packet, and the remainder of the packet is filled with duplicates of some of the data in the packet or in the jitter buffer. As the jitter buffer fills beyond a useful level, the jitter buffer discards progressively larger amounts of data, without necessarily discarding one or more entire packets. This fine-grained management of the amount of data in the jitter buffer maintains a buffer size that can provide a steady stream of packets to the receiver, without significantly impacting the fidelity of a signal represented by the data, and it mitigates the impact of fluctuations in packet inter-arrival times.
    • 基于抖动缓冲区中数据的波动量,抖动缓冲区管理器或多或少主动地节省(R))或丢弃抖动缓冲区中的数据。 抖动缓冲器管理器对整个数据包进行计数,提供,丢弃和/或以其他方式管理各个样本数据单元。 通常,从抖动缓冲器中删除足够的数据以填充接收器的数据包。 然而,如果抖动缓冲区中的数据量较少,则从抖动缓冲区中删除较少的数据并将其放入数据包中,并且数据包的剩余部分将填充数据包中的某些数据或抖动 缓冲。 由于抖动缓冲区超出了有用的级别,抖动缓冲区将逐渐丢弃更大量的数据,而不必丢弃一个或多个整个数据包。 对抖动缓冲器中的数据量的这种细粒度管理保持缓冲器大小,其可以向接收器提供稳定的分组流,而不会显着影响由数据表示的信号的保真度,并且其减轻波动的影响 在包到达时间。
    • 3. 发明授权
    • Distributed telephony resource management method
    • 分布式电话资源管理方法
    • US06477172B1
    • 2002-11-05
    • US09318178
    • 1999-05-25
    • Eric William BurgerAndy Spitzer
    • Eric William BurgerAndy Spitzer
    • H04L1228
    • H04Q3/0016H04L67/42H04Q2213/13107H04Q2213/1338
    • To form communication connections in a set of nodes, the set of nodes is arranged so that at least a subset of the nodes form a ring. Each node in the subset of nodes is coupled to first and second neighboring nodes in the subset by a set of inter-node communication connections. An external communication is received at one of the nodes. The external communication requests access to a communication resource of a specified type. The receiving node determines whether it has an available communication resource of the specified type. If not, the following additional steps are performed. A resource request message is sent to the first and second neighboring nodes of the receiving node to request allocation of a communication resource of the specified type. Each of the first and second neighboring nodes determines whether that node has an available communication resource of the specified type. If not, the neighboring node transmits another resource request message to one of its neighboring nodes other than the node from which the resource request message was received. When any node determines that it has an available communication resource of the specified type, a communications path is formed between the receiving node and the communication resource of the specified type via the inter-node communication connections therebetween.
    • 为了在一组节点中形成通信连接,布置该组节点使得节点的至少一个子集形成环。 节点子集中的每个节点通过一组节点间通信连接耦合到子集中的第一和第二相邻节点。 在一个节点处接收外部通信。 外部通信请求访问指定类型的通信资源。 接收节点确定它是否具有指定类型的可用通信资源。 如果没有,则执行以下附加步骤。 资源请求消息被发送到接收节点的第一和第二相邻节点,以请求分配指定类型的通信资源。 第一和第二相邻节点中的每一个确定该节点是否具有指定类型的可用通信资源。 如果不是,则相邻节点向除了从其接收到资源请求消息的节点之外的其中一个相邻节点发送另一个资源请求消息。 当任何节点确定它具有指定类型的可用通信资源时,通过它们之间的节点间通信连接在接收节点和指定类型的通信资源之间形成通信路径。
    • 4. 发明授权
    • Method for monitoring telephone call progress
    • 监听电话进度的方法
    • US5521967A
    • 1996-05-28
    • US224464
    • 1994-04-07
    • Robert G. NovasAndy Spitzer
    • Robert G. NovasAndy Spitzer
    • H04M1/26H04M1/82H04M11/06H04M11/00
    • H04M1/82H04M1/26H04M11/06
    • A call progress monitor and algorithm for placing a telephone call over the telephone lines. The algorithm is hierarchically arranged having three major routines or portions: tone detection, signal recognition and situation recognition. The tone detection portion of the algorithm measures the power at each of a predetermined number of frequencies of interest. In accordance with one improvement, the data is sampled at a fraction of the rate at which it is supplied, for example, on a T1 channel. The signal recognition portion of the algorithm detects the presence of a particular signal and this stage of the algorithm has been modified to more precisely detect voice. Specifically, power within a voice filter band is compared with power at other predetermined frequencies and a decision is made based on this information. Furthermore, another improved aspect is that data for an entire Epoch is accumulated and stored in a buffer prior to processing as a group. The situation recognition portion of the algorithm determines that a certain sequence or pattern of signals has occurred with a particular timing or cadence.
    • 用于通过电话线拨打电话的呼叫进程监视器和算法。 该算法被分层布置,具有三个主要例程或部分:音调检测,信号识别和情境识别。 该算法的音调检测部分在预定感兴趣频率的每个频率处测量功率。 根据一个改进,数据以其提供速率的一小部分进行采样,例如在T1通道上进行采样。 该算法的信号识别部分检测特定信号的存在,并且该算法的该阶段已被修改以更精确地检测语音。 具体而言,语音滤波器频带内的功率与其他预定频率的功率进行比较,并且基于该信息进行判定。 此外,另一个改进的方面是整个Epoch的数据在作为组进行处理之前被累积并存储在缓冲器中。 该算法的情境识别部分确定特定的序列或信号模式已经发生特定的时间或节奏。
    • 5. 发明授权
    • Method for monitoring telephone call progress
    • 监听电话进度的方法
    • US5325425A
    • 1994-06-28
    • US712538
    • 1991-06-10
    • Robert G. NovasAndy Spitzer
    • Robert G. NovasAndy Spitzer
    • H04M1/26H04M1/82H04M11/06H04M11/00
    • H04M1/82H04M1/26H04M11/06
    • A call progress monitor and algorithm for placing a telephone call over the telephone lines. The algorithm is hierarchically arranged having three major routines or portions: tone detection, signal recognition and situation recognition. The tone detection portion of the algorithm measures the power at each of a predetermined number of frequencies of interest. In accordance with one improvement, the data is sampled at a fraction of the rate at which it is supplied, for example, on a T1 channel. The signal recognition portion of the algorithm detects the presence of a particular signal and this stage of the algorithm has been modified to more precisely detect voice. Specifically, power within a voice filter band is compared with power at other predetermined frequencies and a decision is made based on this information. Furthermore, another improved aspect is that data for an entire Epoch is accumulated and stored in a buffer prior to processing as a group. The situation recognition portion of the algorithm determines that a certain sequence or pattern of signals has occurred with a particular timing or cadence.
    • 用于通过电话线拨打电话的呼叫进程监视器和算法。 该算法被分层布置,具有三个主要例程或部分:音调检测,信号识别和情境识别。 该算法的音调检测部分在预定感兴趣频率的每个频率处测量功率。 根据一个改进,数据以其提供速率的一小部分进行采样,例如在T1通道上进行采样。 该算法的信号识别部分检测特定信号的存在,并且该算法的该阶段已被修改以更精确地检测语音。 具体而言,语音滤波器频带内的功率与其他预定频率的功率进行比较,并且基于该信息进行判定。 此外,另一个改进的方面是整个Epoch的数据在作为组进行处理之前被累积并存储在缓冲器中。 该算法的情境识别部分确定特定的序列或信号模式已经发生特定的时间或节奏。
    • 6. 发明授权
    • Method and apparatus for detecting stuck calls in a communication session
    • 用于在通信会话中检测卡片呼叫的方法和装置
    • US07761577B2
    • 2010-07-20
    • US11047167
    • 2005-01-28
    • Andy Spitzer
    • Andy Spitzer
    • G06F15/16
    • H04L65/1006H04L29/06027H04L43/12H04L65/1083H04L65/608
    • A method and apparatus for terminating a communication session at a first communication device that had established a communication session with a second communication device in response to a determination that the first communication device has not received any media packets from the second communication device within at least one predetermined time period. In the event the first device determines that no media packets have been received from the second device within a first predetermined time period, a probe is sent to the second device. If a positive response is not received by the first device within a second predetermined time period from the sending of the probe, the first device terminates the session. If a positive response is received by the first device in response to the probe, the first device transmits additional probes during the session whenever media packets are not received from the second device within a third predetermined time period and the session is terminated by the first device if a response to the respective probe is not received within the second predetermined time period.
    • 一种在第一通信设备处终止与第二通信设备进行通信会话的方法和设备,其响应于第一通信设备在至少一个第一通信设备中没有从第二通信设备接收到任何媒体分组的确定 预定时间段。 在第一设备确定在第一预定时间段内没有从第二设备接收到媒体分组的情况下,将探测器发送到第二设备。 如果在从发送探测的第二预定时间段内没有被第一设备接收到肯定响应,则第一设备终止会话。 如果第一设备响应于该探测而接收到肯定的响应,则在第三设备期间,当在第三预定时间段内没有从第二设备接收到媒体分组时,第一设备在会话期间发送附加探测,并且该会话由第一设备终止 如果在第二预定时间段内没有接收到对相应探针的响应。
    • 7. 发明授权
    • System and method for constructing phrases for a media server
    • 用于构建媒体服务器的短语的系统和方法
    • US07499863B2
    • 2009-03-03
    • US10141406
    • 2002-05-08
    • Andy Spitzer
    • Andy Spitzer
    • G10L11/00
    • G06F17/30017
    • A system and method for constructing phrases for delivery by a media server over a network to a client. Upon initiation of a session an initial ordered play list comprising a plurality of identifiers is conveyed to the media server. Each identifier on the ordered play list is associated with an audio prompt file constituting a prerecorded audio message, an audio component file comprising a component of a variable audio message to be conveyed to the client or a identifier list comprising at least one identifier. Each identifier contains information sufficient to fetch the content associated with the respective identifier from another server. The media server utilizes the first identifier on the ordered play list to fetch the content associated with that identifier from a server specified by the identifier and removes the respective identifier from the ordered play list. The media server then determines whether the retrieved content is an audio file. If the retrieved content comprises an audio file, the file is played to the client. If the file comprises an identifier list, the identifier list is inserted at the top of the current ordered play list and the media server continues to fetch files associated with the respective identifiers on the ordered play list in sequence. This process continues until no more identifiers are present on the ordered play list. Once the play list has been exhausted, the audio message comprising any prompt files and variable data to be played to the client has been communicated over the network.
    • 一种用于构建用于由媒体服务器通过网络传送到客户端的短语的系统和方法。 在会话开始时,包括多个标识符的初始有序播放列表被传送到媒体服务器。 有序播放列表上的每个标识符与构成预先记录的音频消息的音频提示文件,包括要传送给客户端的可变音频消息的组件的音频组件文件或包括至少一个标识符的标识符列表相关联。 每个标识符包含足以从另一服务器获取与相应标识符相关联的内容的信息。 媒体服务器利用有序播放列表上的第一标识符从由标识符指定的服务器中获取与该标识符相关联的内容,并从有序播放列表中移除相应的标识符。 媒体服务器然后确定所检索的内容是否是音频文件。 如果检索到的内容包括音频文件,则该文件被播放给客户端。 如果文件包括标识符列表,则标识符列表被插入到当前有序播放列表的顶部,并且媒体服务器继续依次获取与有序播放列表上的相应标识符相关联的文件。 该过程继续,直到有序播放列表上不再有标识符。 一旦播放清单已经用尽,包括任何提示文件和要播放到客户端的可变数据的音频消息已经通过网络传送。
    • 8. 发明申请
    • Jitter buffer management
    • 抖动缓冲管理
    • US20050207437A1
    • 2005-09-22
    • US11076815
    • 2005-03-10
    • Andy Spitzer
    • Andy Spitzer
    • H04L12/28H04L12/56H04L29/06
    • H04L47/28H04L12/66H04L29/06027H04L47/2416H04L47/56H04L49/90H04L65/608H04L65/80
    • A sample jitter buffer manager more or less aggressively conserves (rations) or discards data in a jitter buffer, based on the fluctuating amount of data in the jitter buffer. The jitter buffer manager counts, provides, discards and/or otherwise manages individual sample data units, rather than entire packets. Normally, enough data is removed from the jitter buffer to fill a data packet for a receiver. However, if the amount of data in the jitter buffer is low, less data is removed from the jitter buffer and placed into the packet, and the remainder of the packet is filled with duplicates of some of the data in the packet or in the jitter buffer. As the jitter buffer fills beyond a useful level, the jitter buffer discards progressively larger amounts of data, without necessarily discarding one or more entire packets. This fine-grained management of the amount of data in the jitter buffer maintains a buffer size that can provide a steady stream of packets to the receiver, without significantly impacting the fidelity of a signal represented by the data, and it mitigates the impact of fluctuations in packet inter-arrival times.
    • 基于抖动缓冲区中数据的波动量,抖动缓冲区管理器或多或少主动地节省(R))或丢弃抖动缓冲区中的数据。 抖动缓冲器管理器对整个数据包进行计数,提供,丢弃和/或以其他方式管理各个样本数据单元。 通常,从抖动缓冲器中删除足够的数据以填充接收器的数据包。 然而,如果抖动缓冲区中的数据量较少,则从抖动缓冲区中删除较少的数据并将其放入数据包中,并且数据包的剩余部分将填充数据包中的某些数据或抖动 缓冲。 由于抖动缓冲区超出了有用的级别,抖动缓冲区将逐渐丢弃更大量的数据,而不必丢弃一个或多个整个数据包。 对抖动缓冲器中的数据量的这种细粒度管理保持缓冲器大小,其可以向接收器提供稳定的分组流,而不会显着影响由数据表示的信号的保真度,并且其减轻波动的影响 在包到达时间。
    • 9. 发明申请
    • Method and apparatus for detecting stuck calls in a communication session
    • 用于在通信会话中检测卡片呼叫的方法和装置
    • US20050207399A1
    • 2005-09-22
    • US11047167
    • 2005-01-28
    • Andy Spitzer
    • Andy Spitzer
    • H04L12/26H04L12/66H04L29/06
    • H04L65/1006H04L29/06027H04L43/12H04L65/1083H04L65/608
    • A method and apparatus for terminating a communication session at a first communication device that had established a communication session with a second communication device in response to a determination that the first communication device has not received any media packets from the second communication device within at least one predetermined time period. In the event the first device determines that no media packets have been received from the second device within a first predetermined time period, a probe is sent to the second device. If a positive response is not received by the first device within a second predetermined time period from the sending of the probe, the first device terminates the session. If a positive response is received by the first device in response to the probe, the first device transmits additional probes during the session whenever media packets are not received from the second device within a third predetermined time period and the session is terminated by the first device if a response to the respective probe is not received within the second predetermined time period.
    • 一种在第一通信设备处终止与第二通信设备进行通信会话的方法和设备,其响应于第一通信设备在至少一个第一通信设备中没有从第二通信设备接收到任何媒体分组的确定 预定时间段。 在第一设备确定在第一预定时间段内没有从第二设备接收到媒体分组的情况下,将探测器发送到第二设备。 如果在从发送探测的第二预定时间段内没有被第一设备接收到肯定响应,则第一设备终止会话。 如果第一设备响应于该探测而接收到肯定的响应,则在第三设备期间,当在第三预定时间段内没有从第二设备接收到媒体分组时,第一设备在会话期间发送附加探测,并且该会话由第一设备终止 如果在第二预定时间段内没有接收到对相应探针的响应。
    • 10. 发明授权
    • Method for testing large-scale audio conference servers
    • 测试大型音频会议服务器的方法
    • US06888925B2
    • 2005-05-03
    • US10003512
    • 2001-10-26
    • Andy SpitzerRongrong Wu
    • Andy SpitzerRongrong Wu
    • H04M3/24H04M3/32H04M3/56H04Q1/44H04Q1/444H04Q1/45H04M1/24
    • H04M3/56H04M3/24H04M3/323H04Q1/44H04Q1/444H04Q1/45
    • A method and apparatus for testing a conference server. A plurality of test tone signals are generated. In one embodiment each test tone signal comprises the sum of at least two distinct frequency signals. Each test tone signal is employed to simulate a participant and the test tone signals are applied to a plurality of inputs of the conference server. At least some of the test tone signals are combined by the conference server to produce a corresponding plurality of test output signals. The test output signals are analyzed to identify whether the proper test tone signals are present within the respective test output signals. The generation of test tone signals and the analysis of the conference server outputs may be automated to facilitate rapid testing of conference server functionality. An indication of the test results are generated.
    • 一种用于测试会议服务器的方法和装置。 产生多个测试音信号。 在一个实施例中,每个测试音信号包括至少两个不同频率信号的总和。 每个测试音信号用于模拟参与者,测试音信号被应用于会议服务器的多个输入。 至少一些测试音信号由会议服务器组合以产生相应的多个测试输出信号。 分析测试输出信号,以识别相应测试输出信号中是否存在适当的测试音信号。 测试音信号的产生和会议服务器输出的分析可以是自动的,以便于会议服务器功能的快速测试。 产生测试结果的指示。