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194 条记录 (耗时 0.018 秒)

    TRANSCODING AVOIDANCE DURING SINGLE RADIO VOICE CALL CONTINUITY (SRVCC)
    32.
    发明公开
    TRANSCODING AVOIDANCE DURING SINGLE RADIO VOICE CALL CONTINUITY (SRVCC) 审中-公开
    在单个无线电话音呼叫连续性（SRVCC）中对TRANSFODING AVOIDANCE

    公开(公告)号：EP3189644A1

    公开(公告)日：2017-07-12

    申请号：EP15753854.7

    申请日：2015-08-13

    申请人： Intel Corporation

    发明人： SHAN, Chang Hong ,  STOJANOVSKI, Alexandre

    IPC分类号： H04L29/06 ,  H04M7/00 ,  H04W36/00 ,  H04W72/00

    CPC分类号： H04W36/0022 ,  H04L65/1016 ,  H04L65/103 ,  H04L65/80 ,  H04M7/0072 ,  H04M7/1225 ,  H04M2242/04

    摘要： Technology for transcoding avoidance during a single radio voice call continuity (SRVCC) procedure is disclosed. In an example, a mobile switching center (MSC) can include circuitry configured to: receive from a mobility management entity (MME) in a SRVCC packet switch (PS) to circuit switched (CS) request message, selected CODEC information for a selected CODEC used for a user equipment (UE) in an internet protocol (IP) Multimedia Subsystem (IMS) over long term evolution (LTE) system; and communicate the selected CODEC information to a target MSC to enable the target MSC to identify the selected CODEC for the UE to allow the selected CODEC to be used in the CS domain.

    摘要翻译： 披露了用于在单个无线电话音呼叫连续性（SRVCC）过程期间避免转码的技术。 在一个示例中，移动交换中心（MSC）可以包括电路，该电路被配置为：从SRVCC分组交换机（PS）中的移动性管理实体（MME）接收电路交换（CS）请求消息，所选编解码器信息 用于长期演进（LTE）系统中的网际协议（IP）多媒体子系统（IMS）中的用户设备（UE）; 并将选择的CODEC信息传送给目标MSC，以使目标MSC能够识别用于UE的所选CODEC，以允许在CS域中使用所选择的CODEC。

    COMMUNICATION MANAGEMENT SYSTEM, COMMUNICATION SYSTEM, COMPUTER-READABLE RECORDING MEDIUM, AND MAINTENANCE SYSTEM
    33.
    发明公开
    COMMUNICATION MANAGEMENT SYSTEM, COMMUNICATION SYSTEM, COMPUTER-READABLE RECORDING MEDIUM, AND MAINTENANCE SYSTEM 审中-公开
    交通管理系统，通讯系统，记录介质和计算机可读维护系统

    公开(公告)号：EP2798827A4

    公开(公告)日：2015-06-17

    申请号：EP12862181

    申请日：2012-12-26

    申请人： RICOH CO LTD

    发明人： ASAI TAKAHIRO

    IPC分类号： H04M3/00 ,  H04L12/18 ,  H04L29/06 ,  H04M3/56 ,  H04N7/15

    CPC分类号： H04N7/147 ,  H04L12/1813 ,  H04L12/1818 ,  H04L65/605 ,  H04M3/00 ,  H04M3/56 ,  H04M3/567 ,  H04M7/0072 ,  H04N7/142 ,  H04N7/15 ,  H04N7/152

    摘要： A communication management system includes: a storage unit configured to store destination information of a first communication terminal that establishes a first session with a relay device that relays communication data, destination information of a conversion system that performs mutual conversion between communication schemes of communication data transmitted from the first communication terminal and a second communication terminal and establishes a second session with the relay device, and destination information of the second communication terminal that establishes a third session with the conversion system; a receiving unit configured to receive start request information to start communication between the communication terminals from the first communication terminal; an extracting unit configured to extract destination information of each communication terminal and the conversion system stored in the storage unit, based on the received start request information; and a transmitting unit configured to transmit the extracted destination information to the relay device.

    SCALABLE AUDIO IN A MULTI-POINT ENVIRONMENT
    35.
    发明公开
    SCALABLE AUDIO IN A MULTI-POINT ENVIRONMENT 审中-公开
    可伸缩音频播放器在多点区域

    公开(公告)号：EP2502155A4

    公开(公告)日：2013-12-04

    申请号：EP11840587

    申请日：2011-11-11

    申请人： POLYCOM INC

    发明人： FENG JINWEI ,  CHU PETER ,  BOTZKO STEPHEN

    IPC分类号： G06F15/16 ,  G10L19/002 ,  G10L19/02 ,  G10L19/24

    CPC分类号： G10L19/002 ,  G10L19/0212 ,  G10L19/24 ,  G10L25/18 ,  H04M3/568 ,  H04M7/0072

    摘要： Use of a scalable audio codec to implement distributed mixing and/or sender bit rate regulation in a multipoint conference is disclosed. The scalable audio codec allows the audio signal from each endpoint to be split into one or more frequency bands and for the transform coefficients within such bands to be prioritized such that usable audio may be decoded from a subset of the entire signal. The subset may be created by omitting certain frequency bands and/or by omitting certain coefficients within the frequency bands. By providing various rules for each endpoint in a conference, the endpoint can determine the importance of its signal to the conference and can select an appropriate bit rate, thereby conserving bandwidth and/or processing power throughout the conference.

    METHOD, DEVICE, AND SYSTEM FOR CONTROLLING SPEECH ENCODING RATE
    37.
    发明公开
    METHOD, DEVICE, AND SYSTEM FOR CONTROLLING SPEECH ENCODING RATE 审中-公开
    VERFAHREN，VORRICHTUNG系统ZUR STEUERUNG EINER SPRACHKODIERUNGSRATE

    公开(公告)号：EP2635069A1

    公开(公告)日：2013-09-04

    申请号：EP11837577.3

    申请日：2011-11-02

    申请人： Huawei Technologies Co., Ltd.

    发明人： ZHU, Xing ,  GUO, Jiang ,  LIU, Jun ,  YAN, Kai

    IPC分类号： H04W36/36 ,  G10L19/00

    CPC分类号： G10L19/22 ,  H04L65/60 ,  H04M7/0072 ,  H04W4/18 ,  H04W36/36

    摘要： The present invention relates to the communications field and discloses a method, a device, and a system for controlling a voice encoding rate. The method includes: receiving a call pre-handover notification, where the call pre-handover notification carries a voice encoding type used by a local mobile station MS after a call handover; and setting a rate adjustment indication according to the voice encoding type used by the local MS after the call handover, and sending the rate adjustment indication to a peer MS, so that the peer MS performs voice encoding according to the rate adjustment indication until the local MS completes the call handover. The device includes a receiving module, a setting module, and a sending module. In the present invention, a rate adjustment indication is set according to a voice encoding type used by a local MS after a call handover, and the rate adjustment indication is sent to a peer MS, so that the peer MS performs voice encoding according to the rate adjustment indication, thereby avoiding silence caused by a local MS's failure in receiving a high-rate voice frame sent by the peer MS.

    摘要翻译： 本发明涉及通信领域，并且公开了一种用于控制语音编码率的方法，装置和系统。 该方法包括：接收呼叫切换前通知，其中呼叫切换前通知携带在呼叫切换之后由本地移动台MS使用的语音编码类型; 以及根据本地MS在呼叫切换之后使用的语音编码类型设置速率调整指示，并将速率调整指示发送给对等体MS，使得对等体MS根据速率调整指示执行语音编码，直到本地 MS完成呼叫切换。 该设备包括接收模块，设置模块和发送模块。 在本发明中，根据本地MS在呼叫切换之后使用的语音编码类型来设置速率调整指示，并且将速率调整指示发送给对等体MS，使得对等体MS根据 速率调整指示，从而避免本地MS在接收到对等MS发送的高速率语音帧时失败造成的静音。

    Identifying application capabilities for teleconference connections
    39.
    发明公开
    Identifying application capabilities for teleconference connections 失效
    电信技术有限公司电话

    公开(公告)号：EP1585284A3

    公开(公告)日：2012-01-18

    申请号：EP05012593.9

    申请日：1996-02-23

    申请人： APPLE INC.

    发明人： Riddle, Guy G.

    IPC分类号： H04L29/06 ,  H04L12/18

    CPC分类号： H04N7/147 ,  H04L12/1813 ,  H04L12/1818 ,  H04L12/1827 ,  H04L12/1877 ,  H04L29/06 ,  H04L29/06027 ,  H04L51/06 ,  H04L65/1069 ,  H04L65/403 ,  H04L69/24 ,  H04M3/567 ,  H04M7/0072 ,  H04N7/15

    摘要： An automatic method for communicating information, such as teleconference data between teleconferencing systems. A first endpoint (150) identifies communication capabilities to a second endpoint (155) via a first message (1400). The first endpoint notifies the second endpoint of the desire to connect via a second message (1500). The second endpoint notifies the first endpoint of confirmation to connect via a third message (1600). The first and the second endpoints then establish communication according to the communication capabilities. The first and the second endpoints can optimize transfers of teleconference data according to the identified communication capabilities, which is advantageous for the management of merging operations.

    摘要翻译： 用于传送信息的自动方法，例如电话会议系统之间的电话会议数据。 第一端点（150）经由第一消息（1400）识别到第二端点（155）的通信能力。 第一端点通过第二消息通知第二端点连接的需求（1500）。 第二端点通过第三个消息通知第一端点确认连接（1600）。 然后，第一和第二端点根据通信能力建立通信。 第一和第二端点可以根据所识别的通信能力优化电话会议数据的传输，这对于合并操作的管理是有利的。