Measurement of round-trip delay over a network
    1.
    发明申请
    Measurement of round-trip delay over a network 有权
    测量网络上的往返延迟

    公开(公告)号:US20080056154A1

    公开(公告)日:2008-03-06

    申请号:US11516933

    申请日:2006-09-06

    CPC classification number: H04L12/66

    Abstract: In one embodiment, a first audio waveform is produced at a first side of a network connection and then encoded and sent by a first endpoint device to a second endpoint device at a second side of the network connection. A second audio waveform is then detected after being played out by the first endpoint device, the second audio waveform having been produced at the second side of the network connection in response to the second endpoint device playing out the first audio waveform. A round-trip delay is then calculating based on a time period measured from output of the first audio waveform to detection of the second audio waveform. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.

    Abstract translation: 在一个实施例中,在网络连接的第一侧产生第一音频波形,然后由第一端点设备编码并发送到网络连接的第二侧的第二端点设备。 然后在由第一端点设备播放之后检测第二音频波形,响应于第二端点设备播放第一音频波形,已经在网络连接的第二侧产生了第二音频波形。 然后基于从第一音频波形的输出测量到第二音频波形的检测的时间周期来计算往返延迟。 要强调的是,该摘要被提供以符合要求抽象的规则,允许搜索者或其他读者快速确定技术公开内容的主题。

    Non-causal speaker selection for conference multicast
    5.
    发明授权
    Non-causal speaker selection for conference multicast 有权
    会议组播的非因果音箱选择

    公开(公告)号:US06963353B1

    公开(公告)日:2005-11-08

    申请号:US10439147

    申请日:2003-05-14

    Inventor: Scott Firestone

    CPC classification number: H04N7/147 H04N7/152

    Abstract: A method for non-causal speaker selection is provided. In accordance with a particular embodiment of the present invention the method includes receiving a plurality of video streams at a multipoint control unit, each of the plurality of video streams being associated with a respective endpoint of a multipoint conference. A plurality of audio streams may also be received at the multipoint control unit, and each audio stream may be associated with a respective one of the video streams. The audio streams are buffered in respective audio buffers, and the video streams are buffered in respective video buffers. First video data is copied from the video buffers to obtain a low latency video stream for distribution to active conference participants. In a particular embodiment, second video data may be copied from the video buffers to obtain a high latency video stream for distribution to passive conference participants, the high latency video streams being delayed in time with respect to the low latency video stream.

    Abstract translation: 提供了非因果说话者选择的方法。 根据本发明的特定实施例,该方法包括在多点控制单元处接收多个视频流,多个视频流中的每一个与多点会议的相应端点相关联。 也可以在多点控制单元处接收多个音频流,并且每个音频流可以与视频流中的相应一个相关联。 音频流被缓冲在相应的音频缓冲器中,并且视频流被缓冲在相应的视频缓冲器中。 从视频缓冲器复制第一视频数据,以获得低延迟视频流,以便分发给活动的会议参与者。 在特定实施例中,可以从视频缓冲器复制第二视频数据以获得用于分配给被动会议参与者的高等待时间视频流,相对于低等待时间视频流,高等待时间视频流在时间上被延迟。

    Real-time data rate matching across a medium
    6.
    发明授权
    Real-time data rate matching across a medium 失效
    跨媒介的实时数据速率匹配

    公开(公告)号:US06247072B1

    公开(公告)日:2001-06-12

    申请号:US09013866

    申请日:1998-01-27

    Inventor: Scott Firestone

    Abstract: Apparatus and methods for matching data rates is useful for a receiver receiving real-time data over a medium. Implementations feature a process establishing a buffer in a receiver; receiving source data from a source having a nominal source data rate, the received source data arriving at an incoming data rate that differs from time-to-time from the nominal source data rate; filling the buffer with source data as it is received at the incoming data rate and emptying the buffer to provide data for consumption in real time at a consumption data rate; setting a rate-matching factor M, the factor M affecting the rate at which the buffer is emptied; and tracking the level of data in the buffer and resetting the value of M to increase the rate at which the buffer is emptied when the buffer fills above a target range, and resetting the value of M to decrease the rate at which the buffer is emptied when the buffer empties below a target range.

    Abstract translation: 用于匹配数据速率的装置和方法对于在介质上接收实时数据的接收器是有用的。 实现特征在于在接收器中建立缓冲器的过程; 从具有标称源数据速率的源接收源数据,所接收的源数据以与标称源数据速率的时间不同的输入数据速率到达; 在传入数据速率接收到源数据时填充缓冲器,并将缓冲器清空以消耗数据速率实时提供消耗数据; 设置速率匹配因子M,影响缓冲器清空速率的因子M; 并且跟踪缓冲器中的数据级别并重置M的值以在缓冲器填满目标范围时提高缓冲器被清空的速率,并且重置M的值以降低缓冲器被清空的速率 当缓冲液清空到目标范围以下时。

    Measurement of round-trip delay over a network
    7.
    发明授权
    Measurement of round-trip delay over a network 有权
    测量网络上的往返延迟

    公开(公告)号:US07916653B2

    公开(公告)日:2011-03-29

    申请号:US11516933

    申请日:2006-09-06

    CPC classification number: H04L12/66

    Abstract: In one embodiment, a first audio waveform is produced at a first side of a network connection and then encoded and sent by a first endpoint device to a second endpoint device at a second side of the network connection. A second audio waveform is then detected after being played out by the first endpoint device, the second audio waveform having been produced at the second side of the network connection in response to the second endpoint device playing out the first audio waveform. A round-trip delay is then calculating based on a time period measured from output of the first audio waveform to detection of the second audio waveform. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure.

    Abstract translation: 在一个实施例中,在网络连接的第一侧产生第一音频波形,然后由第一端点设备编码并发送到网络连接的第二侧的第二端点设备。 然后在由第一端点设备播放之后检测第二音频波形,响应于第二端点设备播放第一音频波形,已经在网络连接的第二侧产生了第二音频波形。 然后基于从第一音频波形的输出测量到第二音频波形的检测的时间周期来计算往返延迟。 要强调的是,该摘要被提供以符合要求抽象的规则,允许搜索者或其他读者快速确定技术公开内容的主题。

    Lip synchronization for audio/video transmissions over a network
    8.
    发明申请
    Lip synchronization for audio/video transmissions over a network 有权
    通过网络进行音频​​/视频传输的唇形同步

    公开(公告)号:US20080117937A1

    公开(公告)日:2008-05-22

    申请号:US11603849

    申请日:2006-11-22

    Abstract: In one embodiment, a system includes a video mixer coupled with an audio mixer for exchange of information that includes a first set of delay values respecting input audio streams received by the audio mixer from a plurality of source endpoints, and output audio streams sent from the audio mixer to a plurality of destination endpoints. The information further including a second set of delay values respecting the corresponding input video streams. The audio mixer calculates end-to-end video delays, and the video mixer calculates end-to-end audio delays. The audio mixer delays the output audio streams to equalize the end-to-end audio and video delays in the event that the end-to-end audio delays are less than the end-to-end video delays, and the video mixer delays the output video streams to equalize the end-to-end audio and video delays in the event that the end-to-end video delays are less than the end-to-end audio delays.

    Abstract translation: 在一个实施例中,系统包括与音频混合器耦合的视频混合器,用于交换信息,所述信息包括与多个源端点相关的音频混合器接收的输入音频流的第一组延迟值,以及输出从 音频混合器到多个目的地端点。 所述信息还包括相对于相应的输入视频流的第二组延迟值。 音频混合器计算端到端视频延迟,视频混合器可以计算端到端的音频延迟。 在端到端音频延迟小于端到端视频延迟的情况下,音频混合器延迟输出音频流以均衡端到端音频和视频延迟,并且视频混频器延迟 输出视频流以在终端到终端视频延迟小于端到端音频延迟的情况下均衡端对端音频和视频延迟。

    Authenticating an endpoint using a stun server
    9.
    发明申请
    Authenticating an endpoint using a stun server 有权
    使用眩晕服务器验证端点

    公开(公告)号:US20060212702A1

    公开(公告)日:2006-09-21

    申请号:US11087050

    申请日:2005-03-21

    CPC classification number: H04L63/08 H04L9/3271

    Abstract: Authenticating an endpoint using a STUN server includes facilitating a communication session between a first endpoint and a second endpoint over a network. A challenge request is sent to the second endpoint. The challenge request attempts to authenticate the second endpoint and includes an identification. The identification is associated with an expected response identification. A response to the challenge request is received from the second endpoint. The response has an actual response identification. The received response is verified to establish whether the second endpoint is legitimate. The second endpoint is legitimate if the actual response identification includes the expected response identification.

    Abstract translation: 使用STUN服务器认证端点包括促进通过网络的第一端点和第二端点之间的通信会话。 挑战请求被发送到第二端点。 挑战请求尝试认证第二个端点并包含一个标识。 识别与预期响应标识相关联。 从第二端点接收到对质询请求的响应。 响应具有实际的响应标识。 验证接收到的响应以确定第二端点是否合法。 如果实际的响应标识包括预期的响应标识,则第二个端点是合法的。

    System and method for performing distributed video conferencing
    10.
    发明申请
    System and method for performing distributed video conferencing 有权
    用于执行分布式视频会议的系统和方法

    公开(公告)号:US20050078171A1

    公开(公告)日:2005-04-14

    申请号:US10703859

    申请日:2003-11-06

    Abstract: A method for executing a video conference is provided that includes receiving one or more audio streams associated with a video conference from one or more end points and determining an active speaker associated with one of the end points. Audio information associated with the active speaker may be received at one or more media switches. One or more video streams may be suppressed except for a selected video stream associated with the active speaker, the selected video stream propagating to one or more of the media switches during the video conference. The selected video stream may be replicated such that it may be communicated to one or more of the end points associated with a selected one of the media switches.

    Abstract translation: 提供一种用于执行视频会议的方法,其包括从一个或多个端点接收与视频会议相关联的一个或多个音频流,并且确定与所述终点中的一个相关联的活动扬声器。 可以在一个或多个媒体交换机处接收与主动扬声器相关联的音频信息。 除了与活动扬声器相关联的所选视频流之外,可以抑制一个或多个视频流,所选择的视频流在视频会议期间传播到一个或多个媒体交换机。 可以复制所选择的视频流,使得其可以被传送到与所选择的一个媒体交换机相关联的一个或多个终点。

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