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1.发明授权Method and system for managing time-sensitive packetized data streams at a receiver 有权用于在接收机处管理时间敏感的分组化数据流的方法和系统公开(公告)号：US08102766B2
公开(公告)日：2012-01-24
申请号：US11555752
申请日：2006-11-02
CPC分类号： H04J3/0632 , H04Q2213/13034 , H04Q2213/13389
摘要： According to one embodiment of the invention, a method for managing time-sensitive packetized data streams at a receiver includes receiving a time-sensitive packet of a data stream, analyzing an energy level of a payload signal of the packet, and determining whether to drop the packet based on the energy level of the payload signal.
摘要翻译： 根据本发明的一个实施例，一种用于在接收机处理时间敏感分组化数据流的方法包括：接收数据流的时间敏感分组，分析分组的有效载荷信号的能级，并确定是否丢弃 该分组基于有效载荷信号的能级。
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2.发明授权Methods, devices and software for redundant transmission of voice data over a packet network connection established according to an unreliable communication protocol 有权用于通过根据不可靠的通信协议建立的分组网络连接的语音数据的冗余传输的方法，设备和软件公开(公告)号：US07249185B1
公开(公告)日：2007-07-24
申请号：US09702196
申请日：2000-10-30
IPC分类号： G06F15/16
CPC分类号： H04L1/08
摘要： Methods, devices, and software are provided for generating and sending data packets that contain redundant voice data over VoIP connections made under an unreliable network protocol. The redundant data is packaged either in redundant data packets, or in expanded original packets, to repeat data that originally belongs in other packets. Generation of the redundant voice data is either from the transmitting device or from a retransmitting device, such as a router in the network. Generation is triggered either when errors are detected, or simply when the network resources permit it, or both. The received voice data is processed by the second party to the connection, which is typically a telephone call. The redundant voice data that is actually received is discarded. The invention thus ensures that less voice data is lost than in the prior art over VoIP connections made under an unreliable network protocol.
摘要翻译： 提供方法，设备和软件，用于通过不可靠的网络协议进行的VoIP连接生成和发送包含冗余语音数据的数据包。 冗余数据被封装在冗余数据包或扩展的原始数据包中，以重复原本属于其他数据包的数据。 冗余语音数据的生成来自发送设备或来自诸如网络中的路由器的重传设备。 当检测到错误时，或仅当网络资源允许时，或两者都会触发生成。 接收到的语音数据由第二方处理到通常是电话呼叫的连接。 丢弃实际接收到的冗余语音数据。 因此，本发明通过在不可靠的网络协议下进行的VoIP连接来确保比现有技术更少的语音数据丢失。
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公开(公告)号：US07221659B1
公开(公告)日：2007-05-22
申请号：US10039158
申请日：2001-12-31
IPC分类号： H04B3/20
摘要： According to one embodiment of the invention, a method for managing communication impairments between Internet Protocol devices is provided. The method includes determining a transmission of a signal comprising a comfort noise, where the signal is transmitted from a first endpoint to a second endpoint. The method also includes sending a notice signal from the first endpoint to the second endpoint indicating that the signal is transmitted. The method also includes suppressing the signal at the second endpoint in response to the notice signal. According to another embodiment of the invention, a method for managing communication impairments between an Internet Protocol (“IP”) phone and an IP device is provided. The method provides sending a status signal to the device indicating that the phone is operating as a speakerphone. The method also includes suppressing the transmission of any comfort noise to the phone in response to the status signal.
摘要翻译： 根据本发明的一个实施例，提供了一种用于管理因特网协议设备之间的通信损伤的方法。 该方法包括确定包括舒适噪声的信号的传输，其中信号从第一端点传送到第二端点。 该方法还包括从第一端点向第二端点发送指示信号被发送的通知信号。 该方法还包括响应于通知信号来抑制第二端点处的信号。 根据本发明的另一实施例，提供了一种用于管理因特网协议（IP）电话和IP设备之间的通信损害的方法。 该方法提供向设备发送指示电话作为扬声器电话操作的状态信号。 该方法还包括响应于状态信号抑制对手机的任何舒适噪声的传输。
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4.发明授权公开(公告)号：US07136398B1
公开(公告)日：2006-11-14
申请号：US10098270
申请日：2002-03-14
IPC分类号： H04L12/56
CPC分类号： H04L12/1822 , H04L29/06027 , H04L65/1083 , H04L65/403 , H04M3/567
摘要： A method for synchronizing media in a call includes receiving a first input stream of packets of a first media type at a first call resource, generating a first output stream in response to the media streams, receiving synchronization information from a second call resource, and communicating the output stream from the first call resource to a endpoint in synchronization with the second call resource. A video conference bridge includes a first interface operable to receive a video input stream, a processor operable to generate video output, and a second interface operable to receive synchronization information from an audio conference bridge.
摘要翻译： 一种用于在呼叫中同步媒体的方法包括在第一呼叫资源处接收第一媒体类型的分组的第一输入流，响应于媒体流生成第一输出流，从第二呼叫资源接收同步信息，以及通信 与第二呼叫资源同步的从第一呼叫资源到端点的输出流。 视频会议桥包括可操作以接收视频输入流的第一接口，可操作以产生视频输出的处理器和可操作以从音频会议桥接收同步信息的第二接口。
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5.发明授权Method and system for managing call requests in a limited bandwidth environment 有权用于在有限带宽环境中管理呼叫请求的方法和系统公开(公告)号：US07065203B1
公开(公告)日：2006-06-20
申请号：US09850656
申请日：2001-05-07
CPC分类号： H04Q3/0062
摘要： According to one embodiment of the invention, a method for managing call requests in a limited bandwidth environment includes receiving a call request from a user, determining if a voice channel on a communication link is available for the call request, and providing a feedback signal for the user indicating when a voice channel will be available.
摘要翻译： 根据本发明的一个实施例，一种用于在有限带宽环境中管理呼叫请求的方法包括从用户接收呼叫请求，确定通信链路上的语音信道是否可用于呼叫请求，以及提供用于 用户指示语音信道何时可用。
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6.发明授权公开(公告)号：US06853716B1
公开(公告)日：2005-02-08
申请号：US09836055
申请日：2001-04-16
申请人： Shmuel Shaffer , Michael E. Knappe
发明人： Shmuel Shaffer , Michael E. Knappe
CPC分类号： H04M3/56 , G10L17/00 , H04L63/0861 , H04M3/568 , H04M3/569 , H04M2201/41
摘要： A system and method for identifying a participant during a conference call include the capability to receive a packet containing data that represents audible sounds spoken by one of a plurality of participants in a conference call and to determine a speaker of the audible sounds using voice profile information of the participants. The system and method further include the capability to provide identification information of the speaker to the other participants in the conference call contemporaneously with providing audible sounds based on the data to those participants.
摘要翻译： 用于在电话会议期间识别参与者的系统和方法包括接收包含数据的分组的能力，所述数据包含表示在电话会议中的多个参与者之一所讲的可听见的声音，并且使用语音简档信息确定可听见的声音的扬声器 的参与者。 该系统和方法还包括能够向会议呼叫中的其他参与者提供扬声器的识别信息，同时基于该数据向那些参与者提供可听见的声音。
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7.发明授权Spatial audio processing method and apparatus for context switching between telephony applications 失效用于电话应用之间的上下文切换的空间音频处理方法和装置
公开(公告)号：US6011851A
公开(公告)日：2000-01-04
申请号：US880484
申请日：1997-06-23
CPC分类号： H04S1/002 , H04S2400/11 , H04S5/00 , H04S7/30
摘要： Multiple audio streams are spatially separated with a context switching system to allow a listener to mentally focus on individual point sources of auditory information in the presence of other sound sources. The switching system simultaneously directs incoming sound sources to different spatial processors. Each spatial processor moves the received sound sources to different audibly perceived point sources. The outputs from the spatial processors are mixed into a stereo signal with left and right outputs and then output to the listener. Important sound sources are moved to a foreground point source for increased intelligibility while less important source sources are moved to a background point source.
摘要翻译： 多个音频流与上下文切换系统在空间上分离，以允许听众在存在其他声源的情况下精神上集中于听觉信息的各个点源。 交换系统同时将传入的声源引导到不同的空间处理器。 每个空间处理器将接收的声源移动到不同的可听觉感知的点源。 来自空间处理器的输出被混合成具有左和右输出的立体声信号，然后输出到听众。 重要的声源被移动到前景点源以增加可懂度，而较不重要的源数据源被移动到背景点源。
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公开(公告)号：US07194084B2
公开(公告)日：2007-03-20
申请号：US11239542
申请日：2005-09-28
申请人： Shmuel Shaffer , Michael E. Knappe
发明人： Shmuel Shaffer , Michael E. Knappe
IPC分类号： H04M9/08
CPC分类号： H04R27/00 , G10L19/008
摘要： Systems and methods are disclosed for packet voice conferencing. An encoding system accepts two sound field signals, representing the same sound field sampled at two spatially-separated points. The relative delay between the two sound field signals is detected over a given time interval. The sound field signals are combined and then encoded as a single audio signal, e.g., by a method suitable for monophonic VoIP. The encoded audio payload and the relative delay are placed in one or more packets and sent to a decoding device via the packet network.The decoding device uses the relative delay to drive a playout splitter—once the encoded audio payload has been decoded, the playout splitter creates multiple presentation channels by inserting the transmitted relative delay in the decoded signal for one (or more) of the presentation channels. The listener thus perceives a speaker's voice as originating from a location related to the speaker's physical position at the other end of the conference. An advantage of these embodiments is that a pseudo-stereo conference can be conducted with virtually the same bandwidth as a monophonic conference.
摘要翻译： 公开了用于分组语音会议的系统和方法。 编码系统接收两个声场信号，表示在两个空间分离点采样的相同声场。 在给定的时间间隔内检测两个声场信号之间的相对延迟。 声场信号被组合，然后被编码为单个音频信号，例如通过适合于单声道VoIP的方法。 编码的音频有效载荷和相对延迟被放置在一个或多个分组中，并且经由分组网络被发送到解码装置。 解码装置使用相对延迟来驱动播出分离器 - 一旦编码的音频有效载荷已被解码，播放分离器通过将一个（或更多个）演示频道的解码信号中插入传输的相对延迟来创建多个显示频道。 因此，听众将演讲者的声音从与会议另一端的演讲人的身体位置相关的位置发出。 这些实施例的优点是可以以与单声道会议几乎相同的带宽进行伪立体会议。
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公开(公告)号：US06850496B1
公开(公告)日：2005-02-01
申请号：US09591891
申请日：2000-06-09
申请人： Michael E. Knappe , Shmuel Shaffer
发明人： Michael E. Knappe , Shmuel Shaffer
CPC分类号： H04L65/403 , H04L65/4038 , H04M3/56 , H04M3/568 , H04M7/006
摘要： A system and method are disclosed for packet voice conferencing. The system and method divide a conferencing presentation sound field into sectors, and allocate one or more sectors to each conferencing endpoint. At some point between capture and playout, the voice data from each endpoint is mapped into its designated sector or sectors. Thereafter, when the voice data from a plurality of participants from multiple endpoints is combined, a listener can identify a unique apparent location within the presentation sound field for each participant. The system allows a conference participant to increase their comprehension when multiple participants speak simultaneously, as well as alleviate confusion as to who is speaking at any given time.
摘要翻译： 公开了用于分组语音会议的系统和方法。 系统和方法将会议演示声场分成扇区，并为每个会议端点分配一个或多个扇区。 在捕获和播放之间的某个时刻，来自每个端点的语音数据被映射到其指定的扇区或扇区中。 此后，当来自多个端点的多个参与者的语音数据被组合时，收听者可以为每个参与者识别呈现声场内的唯一表观位置。 该系统允许会议参与者在多个参与者同时进行演讲时增加理解能力，并减轻在任何时候发言者的混淆。
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10.发明授权Telephony-enabled network processing device with separate TDM bus and host system backplane bus 失效具有独立TDM总线和主机系统背板总线的支持电话功能的网络处理设备公开(公告)号：US06240084B1
公开(公告)日：2001-05-29
申请号：US08729245
申请日：1996-10-10
IPC分类号： H04L1266
CPC分类号： H04L12/66
摘要： A PC-based server platform includes a first backplane bus used for transferring data and commands to various PC peripheral devices. A network router and a telephony endpoint card are coupled to the backplane bus and separately coupled through a second Time Division Multiplexed (TDM) bus. The router includes interfaces to various packet switched networks such as a Wide Area Network (WAN) and a Local Area Network (LAN). The TDM bus is used to route telephony data between the different Internet Protocol (IP)-based networks and the telephony card independently of the host system. The PC host processor also uses the router as a standard LAN interface for transferring data packets. A DSP voice processing card is coupled between the backplane bus and the TDM bus to compress and decompress the telephony data transferred on the TDM bus.
摘要翻译： 基于PC的服务器平台包括用于将数据和命令传送到各种PC外围设备的第一个背板总线。 网络路由器和电话端点卡耦合到背板总线，并通过第二时分复用（TDM）总线单独耦合。 路由器包括各种分组交换网络的接口，例如广域网（WAN）和局域网（LAN）。 TDM总线用于在不同的基于互联网协议（IP）的网络和独立于主机系统的电话卡之间路由电话数据。 PC主机处理器还使用路由器作为传输数据包的标准LAN接口。 DSP语音处理卡耦合在背板总线和TDM总线之间，以压缩和解压缩TDM总线上传送的电话数据。
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