Scalable voice over IP system providing independent call bridging for outbound calls initiated by user interface applications
    213.
    发明授权
    Scalable voice over IP system providing independent call bridging for outbound calls initiated by user interface applications 有权
    可扩展的IP语音系统,为用户界面应用程序发起的出站呼叫提供独立的呼叫桥接

    公开(公告)号:US06930999B1

    公开(公告)日:2005-08-16

    申请号:US09606692

    申请日:2000-06-30

    Abstract: An IP telephony gateway and a user interface resource enable a subscriber to place an outgoing call according to the voice over IP (H.323) protocol to a destination party from a user interface session of an intelligent dial tone service such as voice activated dialing, and resume the user interface session upon completion of the outgoing call with the destination party. The IP telephony gateway establishes a user interface session for the subscriber with the user interface resource across a first Real Time Protocol (RTP) data stream. The user interface resource initiates a second RTP data stream to a destination party in response to reception of a prescribed command from the subscriber. Although an RTP bridge connecting the first and second RTP data streams can be maintained by the user interface resource, the user interface resource may also use the Empty Capability Set feature in the H.323 standard to cause the IP telephony gateway to close the first and second RTP data streams to the user interface resource. The user interface resource then issues Non-Empty Capability Set messages to the IP telephony gateway for the first and second RTP data streams, causing the IP telephony gateway to internally bridge the first and second RTP data streams. The user interface resource monitors connections between the subscriber and the destination party, and upon detecting a disconnect by the destination party causes the IP telephony gateway to resume the user interface session, by repeating the sequence of sending Empty Capability Set and Non-Empty Capability Set messages to the IP telephony gateway to break down the bridge and re-establish the connection between the subscriber and the user interface resource.

    Abstract translation: IP电话网关和用户接口资源使订户能够根据IP语音(H.323)协议将出呼叫从智能拨号音服务的用户接口会话(例如语音激活拨号)发送到目的地, 并且在与目标方进行呼出完成时恢复用户界面会话。 IP电话网关通过第一实时协议(RTP)数据流为用户建立与用户接口资源的用户接口会话。 响应于从订户接收到规定的命令,用户接口资源向目的地发起第二RTP数据流。 虽然连接第一和第二RTP数据流的RTP桥可以由用户界面资源来维护,但是用户接口资源也可以使用H.323标准中的空能力集特征来使得IP电话网关关闭第一和第 第二RTP数据流到用户界面资源。 然后,用户接口资源向第一和第二RTP数据流的IP电话网关发出非空能力设置消息,导致IP电话网关内部桥接第一和第二RTP数据流。 用户接口资源监视用户和目的方之间的连接,并且在检测到目的方之间的断开连接时,通过重复发送空能力集和非空能力集的顺序使得IP电话网关恢复用户界面会话 消息到IP电话网关,以分解网桥并重新建立用户和用户界面资源之间的连接。

    Communication system
    215.
    发明授权
    Communication system 有权
    通讯系统

    公开(公告)号:US06901240B2

    公开(公告)日:2005-05-31

    申请号:US10027643

    申请日:2001-12-20

    Applicant: Mario Tokoro

    Inventor: Mario Tokoro

    Abstract: When a line for communication of audio signals between first and second portable-telephone devices through a first portable-telephone base station, a second portable-telephone base station and a portable-telephone line network has been connected, IP addresses are exchanged. The IP addresses are required for communicating image data by first and second terminal adapters associated with the first and second portable-telephone devices respectively through an OCN. Image data is then transmitted to the IP addresses received in the communication through the OCN. To put it concretely, an image of the user of the first portable-telephone device is transmitted by the first terminal adapter through the OCN to the IP address of the second terminal adapter to be displayed by a second television receiver associated with the second terminal adapter. By the same token, an image of the user of the second portable-telephone device is transmitted by the second terminal adapter through the OCN to the IP address of the first terminal adapter to be displayed by a first television receiver associated with the first terminal adapter. Thus, by merely taking a portable-telephone device to a place like a store selling home electrical appliances including a television.

    Abstract translation: 当通过第一便携式电话基站,第二便携式电话基站和便携式电话线路网络连接第一和第二便携式电话设备之间的音频信号的线路时,交换IP地址。 需要IP地址来分别通过OCN与第一和第二便携式电话设备相关联的第一和第二终端适配器通信图像数据。 然后,图像数据通过OCN发送到通信中接收到的IP地址。 具体地说,第一便携式电话设备的用户的图像由第一终端适配器通过OCN发送到第二终端适配器的IP地址,以由与第二终端适配器相关联的第二电视接收机显示 。 同样,第二便携式电话设备的用户的图像由第二终端适配器通过OCN发送到第一终端适配器的IP地址,以由与第一终端适配器相关联的第一电视接收机显示 。 因此,通过仅将便携式电话设备带到诸如出售包括电视在内的家用电器的商店的地方。

    Method and apparatus for functional architecture of voice-over-IP SIP network border element
    218.
    发明申请
    Method and apparatus for functional architecture of voice-over-IP SIP network border element 有权
    IP语音SIP网络边界元素功能架构的方法和装置

    公开(公告)号:US20050083912A1

    公开(公告)日:2005-04-21

    申请号:US10790264

    申请日:2004-03-01

    Abstract: In order to provide a single common cost-efficient architecture for real time communication services for audio, video, and data over internet protocol, a voice over internet protocol (VoIP) system and architecture is provided by placing border elements (BEs) at the interface boundaries between the access network the user devices use and the VoIP infrastructure. The BEs use SIP protocol as the access call control protocol over any access networking technologies, for example, IP, Ethernet, ATM, and FR, and provides all services transparently to the end users that use SIP-enabled devices. To enable a scalable system, the SIP BEs are decomposed into separate communicating entities that make the SIP BE scalable and provide new capabilities not previously available by a self-contained SIP BE. Further, multiple levels of decomposition of a SIP BE can be provided by the present invention supporting a flexible and scalable SIP BE design that further improves system efficiencies and cost advantages as compared to use of single integrated border or edge elements. Further, a scalable SIP BE, made up of a plurality of physical entities for optimization of a large scale design, acts as a single integrated functional entity to logically execute a set of functions at the border of a VoIP infrastructure.

    Abstract translation: 为了提供用于通过因特网协议的音频,视频和数据的实时通信服务的单一通用的成本有效的架构,通过在接口上放置边界元素(BE)来提供语音网络协议(VoIP)系统和架构 用户设备使用的接入网络与VoIP基础设施之间的边界。 BE通过任何接入网络技术(例如IP,以太网,ATM和FR)使用SIP协议作为接入呼叫控制协议,并向使用支持SIP的设备的最终用户透明地提供所有业务。 为了实现可扩展的系统,SIP BE被分解成单独的通信实体,使得SIP可扩展,并且提供由独立SIP BE以前不可用的新功能。 此外,本发明可以提供支持灵活且可扩展的SIP BE设计的SIP BE的多个级别的分解,与使用单个集成边界或边缘元素相比,其进一步提高系统效率和成本优势。 此外,由用于优化大规模设计的多个物理实体组成的可扩展SIP BE充当单个集成功能实体,以在VoIP基础设施的边界处逻辑地执行一组功能。

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