Devices, softwares and methods for predicting reconstruction of encoded frames and for adjusting playout delay of jitter buffer
    12.
    发明授权
    Devices, softwares and methods for predicting reconstruction of encoded frames and for adjusting playout delay of jitter buffer 有权
    用于预测编码帧重建的设备,软件和方法,以及调整抖动缓冲区的播放延迟

    公开(公告)号:US07251241B1

    公开(公告)日:2007-07-31

    申请号:US10225814

    申请日:2002-08-21

    Abstract: Source devices, softwares and methods are provided for predicting at a transmitting source how well a specific encoded frame, relative to other frames in a stream, would be reconstructed if it were not available for eventual playout at a receiving destination. If poorly, then the specific encoded frame is packetized and transmitted advantageously with respect to other packets in a stream. The transmitted stream will have higher jitter. Predicting is performed at the source by synthesizing a replicated frame, which emulates how the specific frame would be reconstructed at the destination. Then the replicated frame is compared with the decoded specific frame. The eventual playout will have a higher fidelity to the original, since reconstructed from more critical frames. Receiving devices, softwares and methods are provided for receiving in a jitter buffer a stream of packets containing encoded data frames, determining that they have been packetized differently from each other, and setting a playout delay of the jitter buffer responsive to the determination, regardless of high jitter.

    Abstract translation: 提供源设备,软件和方法用于在发送源处预测如果在接收目的地不可用于最终播放时,相对于流中的其他帧将被重建的特定编码帧将如何被重建。 如果不好,那么特定编码的帧被打包并相对于流中的其他分组有利地发送。 传输的流将具有更高的抖动。 通过合成复制帧在源处执行预测,仿真帧模拟如何在目的地重建特定帧。 然后将复制的帧与解码的特定帧进行比较。 最终的播出将具有更高的保真度,原始的,因为从更关键的帧重构。 提供接收设备,软件和方法,用于在抖动缓冲器中接收包含编码数据帧的分组流,确定它们已经彼此不同地打包,并且响应于该确定设置抖动缓冲器的播出延迟,而不管 高抖动

    System and method for performing distributed multipoint video conferencing
    13.
    发明申请
    System and method for performing distributed multipoint video conferencing 有权
    用于执行分布式多点视频会议的系统和方法

    公开(公告)号:US20070064901A1

    公开(公告)日:2007-03-22

    申请号:US11210325

    申请日:2005-08-24

    CPC classification number: H04N7/15 H04N7/152

    Abstract: According to an embodiment of the present invention, an apparatus for performing video conferencing is provided that includes an I-frame injector element operable to intercept I-frame requests from one or more end points and to attempt to service the I-frame requests such that at least a portion of the requests are prevented from propagating back to an originating sender. In more specific embodiments, when a receiver endpoint sends a fast video update (FVU) request upstream, it is intercepted by the I-frame injector element and rather than passing the FVU request to the sender the I-frame injector element replaces a next P-frame from the sender with an I-frame, whereby the I-frame is constructed so that when decoded, it matches the P-frame that it replaced. In still more detailed embodiments, the I-frame injector element operates in one of three modes that are associated with bandwidth parameters.

    Abstract translation: 根据本发明的实施例,提供了一种用于执行视频会议的装置,其包括I帧注入器元件,其可操作以截取来自一个或多个端点的I帧请求并尝试对I帧请求进行服务,使得 至少一部分请求被阻止传播回始发端。 在更具体的实施例中,当接收机端点在上游发送快速视频更新(FVU)请求时,它被I帧注入器元件拦截,而不是将FVU请求传递给发送器,I帧注入器元件替换下一个P - 帧从具有I帧的发送器发送,由此构造I帧,使得当被解码时,它与其所取代的P帧匹配。 在更详细的实施例中,I帧注入器元件以与带宽参数相关联的三种模式之一操作。

    Method and apparatus for reconstructing voice information
    14.
    发明申请
    Method and apparatus for reconstructing voice information 有权
    重建语音信息的方法和装置

    公开(公告)号:US20060122835A1

    公开(公告)日:2006-06-08

    申请号:US11336742

    申请日:2006-01-19

    CPC classification number: G10L19/005

    Abstract: A communication system includes a destination that receives voice samples and a voice parameter generated by a source. The destination uses the voice samples and voice parameter to reconstruct voice information in response to a packet loss. The destination may reconstruct voice information from multiple sources.

    Abstract translation: 通信系统包括接收语音样本的目的地和由源产生的语音参数。 目的地使用语音样本和语音参数来重构语音信息以响应分组丢失。 目的地可以从多个来源重建语音信息。

    Techniques for providing a conference with a virtual participant
    16.
    发明授权
    Techniques for providing a conference with a virtual participant 有权
    为虚拟参与者提供会议的技术

    公开(公告)号:US07574473B2

    公开(公告)日:2009-08-11

    申请号:US11417804

    申请日:2006-05-03

    Applicant: Luke Surazski

    Inventor: Luke Surazski

    CPC classification number: G06Q10/10

    Abstract: Techniques for conducting a meeting using a meeting management system are provided. A connection from a user to a connection interface is received for a meeting being offered by the meeting management system. An input is received that provides contact information for a user. The user may choose to be a virtual participant in the meeting. A meeting interface is provided during the meeting using the meeting management system where the interface includes an option to contact the user. The user is listed as a virtual participant in the meeting. A request to contact the user is received at the meeting interface and the user is automatically connected to the user using the contact information.

    Abstract translation: 提供了使用会议管理系统进行会议的技术。 接收从会议管理系统提供的会议从用户到连接接口的连接。 接收到为用户提供联系人信息的输入。 用户可以选择在会议中成为虚拟参与者。 在会议期间使用会议管理系统提供会议界面,其中界面包括与用户联系的选项。 该用户被列为会议中的虚拟参与者。 在会议界面接收到与用户联系的请求,并且使用联系人信息自动地将用户连接到用户。

    Network resource optimization in a video conference
    17.
    发明申请
    Network resource optimization in a video conference 审中-公开
    视频会议中的网络资源优化

    公开(公告)号:US20070263824A1

    公开(公告)日:2007-11-15

    申请号:US11406112

    申请日:2006-04-18

    CPC classification number: H04N7/152 H04L12/1827 H04L65/4038 H04M3/567

    Abstract: A conferencing system includes a mixer operable to process audio and video packets received from a plurality of endpoint devices, and to transmit a processed audio/video stream back to the endpoint devices. A server connects with the mixer and the endpoint devices. The server is operable to send a first transmission to a first endpoint device that causes the first endpoint device to stop transmitting video packets to the mixer. The server is further operable to send a second transmission to a second endpoint device that causes the second endpoint to start transmitting video packets to the mixer. 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. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims. 37 CFR 1.72(b).

    Abstract translation: 会议系统包括可操作以处理从多个端点设备接收的音频和视频分组的混合器,以及将经处理的音频/视频流发送回端点设备。 服务器与混音器和端点设备连接。 服务器可操作地向第一端点设备发送第一传输,其使得第一端点设备停止向混合器传输视频分组。 服务器还可操作以将第二传输发送到第二端点设备,其使得第二端点开始向混合器发送视频分组。 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。 提交它的理解是,它不会用于解释或限制权利要求的范围或含义。 37 CFR 1.72(b)。

    Devices, software and methods for allocating jitter buffer memory based on network region of network endpoint
    18.
    发明授权
    Devices, software and methods for allocating jitter buffer memory based on network region of network endpoint 有权
    基于网络端点网络区域分配抖动缓冲存储器的设备,软件和方法

    公开(公告)号:US07277943B1

    公开(公告)日:2007-10-02

    申请号:US09832422

    申请日:2001-04-10

    Applicant: Luke Surazski

    Inventor: Luke Surazski

    Abstract: Devices, software and methods are provided for allocating different amounts of memory of a network communication device as jitter buffer for different channels, such as VoIP connections. The channels are classified according to regions of their endpoints in the network. Jitter statistics are tracked for connections with endpoints in each region. The statistics are used to determine how much memory to allocate as a jitter buffer for each channel in a subsequent connection, depending on the region of the endpoint.

    Abstract translation: 提供了设备,软件和方法,用于将不同数量的网络通信设备的存储器分配为诸如VoIP连接的不同信道的抖动缓冲器。 信道根据其在网络中的端点的区域进行分类。 跟踪每个区域中与端点连接的抖动统计信息。 统计信息用于确定根据端点的区域,为后续连接中每个通道的抖动缓冲区分配多少内存。

    Method and System for Call Answer While Connected to Voice Mail
    19.
    发明申请
    Method and System for Call Answer While Connected to Voice Mail 有权
    连接语音邮件时呼叫应答的方法和系统

    公开(公告)号:US20060177020A1

    公开(公告)日:2006-08-10

    申请号:US11278611

    申请日:2006-04-04

    CPC classification number: H04M3/53366 H04M3/428 H04M7/006

    Abstract: A method and system for call answer while connected to voice mail includes generating real-time packets for transmission of a message toward a voice mail system. Generation of the real-time packets is interrupted upon a call answer by a party generating the message. Generation of the real-time packets for transmission of the message toward the voice mail system is resumed after a delay associated with the call answer. An indication is provided to the voice mail system that the real-time packets generated subsequent to the delay follow the real-time packets generated prior to the delay in the message by a time amount less than the delay.

    Abstract translation: 一种连接语音邮件的呼叫应答方法和系统包括生成用于向语音邮件系统发送消息的实时数据包。 通过生成消息的一方的呼叫应答来中断实时分组的生成。 在与呼叫应答相关联的延迟之后,恢复生成用于向语音邮件系统发送消息的实时分组。 向话音邮件系统提供指示,在延迟之后生成的实时分组遵循在消息中延迟之前生成的实时分组小于延迟的时间量。

    Method and apparatus for transitioning comfort noise in an IP-based telephony system
    20.
    发明授权
    Method and apparatus for transitioning comfort noise in an IP-based telephony system 有权
    用于在基于IP的电话系统中转换舒适噪声的方法和装置

    公开(公告)号:US06873604B1

    公开(公告)日:2005-03-29

    申请号:US09629032

    申请日:2000-07-31

    CPC classification number: H04M7/0072 G10L19/012 H04M7/006

    Abstract: A method is disclosed, where the RX unit of an IP telephone waits for a first silence indication packet to be received. When the first silence indication packet is received, the RX begins attenuating incoming packets. The rate of attenuation may occur at a rate responsive to an indication provided by the TX unit. The RX unit also mixes comfort noise with the attenuated incoming packets. The RX unit the waits for a second silence indication packet to be received. When the second silence indication is received the RX unit mixes in comfort noise at a level responsive to an indication provided by the TX unit. The RX units then waits for voice packets to be received from the TX unit. When normal voice activity resumes, the RX unit stops the generation of comfort noise and begins playing voice packets again.

    Abstract translation: 公开了一种方法,其中IP电话的RX单元等待接收到第一沉默指示分组。 当接收到第一个静音指示包时,RX开始衰减传入的数据包。 衰减速率可以以响应于由TX单元提供的指示的速率发生。 RX单元还将舒适噪声与衰减的传入数据包混合。 RX单元等待接收到第二个静默指示包。 当接收到第二个静音指示时,RX单元以响应于TX单元提供的指示的电平混合舒适噪声。 然后,RX单元等待从TX单元接收到语音分组。 当恢复正常的语音活动时,RX单元停止生成舒适噪音,并再次开始播放语音数据包。

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