公开(公告)号：US20080130511A1

公开(公告)日：2008-06-05

申请号：US11930678

申请日：2007-10-31

申请人： Ki-Jong Koo ,  Dae-Ho Kim ,  Tae-Gyu Kang ,  Do Young Kim ,  Hae Won Jung

发明人： Ki-Jong Koo ,  Dae-Ho Kim ,  Tae-Gyu Kang ,  Do Young Kim ,  Hae Won Jung

IPC分类号： G06F11/30

CPC分类号： H04L47/801 ,  G10L19/24 ,  H04L43/0829 ,  H04L43/0852 ,  H04L47/70 ,  H04L47/822

摘要： A method and apparatus for controlling a variable bit-rate voice codec are provided. The method of controlling the variable bit-rate voice codec may include: extracting calling capability of terminals that request a new call to be made; extracting network parameters from existing calls in the network through an exchanged packet; measuring voice quality of the existing calls based on the extracted network parameters; and determining whether to permit a new call to be made based on the measured voice quality and the calling capability. Accordingly, it is possible to secure QoS of a voice service between terminals by adjusting a transmission rate of the variable bit-rate codec based on transmission capability of a network.

摘要翻译： 提供了一种用于控制可变比特率语音编解码器的方法和装置。 控制可变比特率语音编解码器的方法可以包括：提取请求新呼叫的终端的呼叫能力; 通过交换的分组从网络中的现有呼叫提取网络参数; 基于提取的网络参数测量现有呼叫的语音质量; 以及基于测量的语音质量和呼叫能力来确定是否允许进行新的呼叫。 因此，可以通过基于网络的传输能力调整可变比特率编解码器的传输速率来确保终端之间的语音服务的QoS。

公开(公告)号：US07907523B2

公开(公告)日：2011-03-15

申请号：US11930678

申请日：2007-10-31

申请人： Ki-Jong Koo ,  Dae-Ho Kim ,  Tae-Gyu Kang ,  Do Young Kim ,  Hae Won Jung

发明人： Ki-Jong Koo ,  Dae-Ho Kim ,  Tae-Gyu Kang ,  Do Young Kim ,  Hae Won Jung

IPC分类号： H04L12/26

CPC分类号： H04L47/801 ,  G10L19/24 ,  H04L43/0829 ,  H04L43/0852 ,  H04L47/70 ,  H04L47/822

摘要： A method and apparatus for controlling a variable bit-rate voice codec are provided. The method of controlling the variable bit-rate voice codec may include: extracting calling capability of terminals that request a new call to be made; extracting network parameters from existing calls in the network through an exchanged packet; measuring voice quality of the existing calls based on the extracted network parameters; and determining whether to permit a new call to be made based on the measured voice quality and the calling capability. Accordingly, it is possible to secure QoS of a voice service between terminals by adjusting a transmission rate of the variable bit-rate codec based on transmission capability of a network.

摘要翻译： 提供了一种用于控制可变比特率语音编解码器的方法和装置。 控制可变比特率语音编解码器的方法可以包括：提取请求新呼叫的终端的呼叫能力; 通过交换的分组从网络中的现有呼叫提取网络参数; 基于所提取的网络参数来测量现有呼叫的语音质量; 以及基于测量的语音质量和呼叫能力来确定是否允许进行新的呼叫。 因此，可以通过基于网络的传输能力调整可变比特率编解码器的传输速率来确保终端之间的语音服务的QoS。

公开(公告)号：US20080081648A1

公开(公告)日：2008-04-03

申请号：US11702275

申请日：2007-02-05

申请人： Tae Gyu Kang ,  Ki Jong Koo ,  Dae Ho Kim ,  Do Young Kim ,  Hae Won Jung

发明人： Tae Gyu Kang ,  Ki Jong Koo ,  Dae Ho Kim ,  Do Young Kim ,  Hae Won Jung

IPC分类号： H04B7/00

CPC分类号： H04M7/0072 ,  H04L65/1069 ,  H04L65/80

摘要： Provided is a method and apparatus for minimizing the number of transcodings between network devices in a multi-network multi-codec environment. The method includes: creating a received codec list by receiving a transmit codec comprised in a call setting message from a transmission device and the number of transcodings of the transmit codec, which has been performed from a codec of an initial transmission device; creating a codec Quality of Service (QoS) list containing total codecs of the multi-network and quality information of each of the total codecs; creating a transcodec list containing internally providable transcodecs and quality information of the transcodecs based on the codec QoS list; and creating an updated codec list by adding codecs of the transcodec list matching codecs of the received codec list to the received codec list and adjusting codec priority according to the number of transcodings. Accordingly, since the number of transcodings can be minimized, a quality decrease of original media can be minimized.

摘要翻译： 提供了一种用于最小化多网络多编解码器环境中的网络设备之间的转码次数的方法和装置。 该方法包括：通过从传输设备接收包括在呼叫设置消息中的传输编解码器和已经从初始传输设备的编解码器执行的传输编解码器的转码次数来创建接收的编解码器列表; 创建包含多网络的总编解码器和每个编解码器的质量信息的编解码器服务质量（QoS）列表; 基于编解码器QoS列表创建包含内容可提供的转录代码和转录代码的质量信息的转录列表; 以及通过将接收到的编解码器列表的编解码器匹配的转码器列表的编解码器添加到接收到的编解码器列表并根据转码数量来调整编解码器优先级来创建更新的编解码器列表。 因此，由于能够最小化转码次数，因此可以使原始媒体的质量下降最小化。

公开(公告)号：US08117029B2

公开(公告)日：2012-02-14

申请号：US11928557

申请日：2007-10-30

申请人： Dae-Ho Kim ,  Tae-Gyu Kang ,  Ki-Jong Koo ,  Do Young Kim ,  Hae Won Jung

发明人： Dae-Ho Kim ,  Tae-Gyu Kang ,  Ki-Jong Koo ,  Do Young Kim ,  Hae Won Jung

IPC分类号： G10L21/00

CPC分类号： G10L25/69 ,  G10L19/18

摘要： Provided is a transmission apparatus for matching sound quality measurement sections of a variable bandwidth multi-codec. The apparatus includes a measurement section setting unit setting a measurement section, which is to be measured for sound quality, in units of time; a first conversion unit converting the measurement section into a measurement section in units of samples; and an information synthesis unit synthesizing information regarding the measurement section in units of samples with a digital original sound and outputting the synthesis result. In addition, provided is a method of matching a measurement section of a reference sound, based on which the end-to-end sound quality measurement of the variable bandwidth multi-codec is performed, and a measurement section of a sound produced by the variable bandwidth multi-codec in a real-time Internet multimedia service. Therefore, distortion of measurement results due to un-matching measurement sections can be reduced.

摘要翻译： 提供了一种用于匹配可变带宽多编解码器的音质测量部分的发送装置。 该装置包括以时间为单位设置要测量声音质量的测量部分的测量部分设置单元; 第一转换单元，以样本为单位将测量部转换为测量部; 以及信息合成单元，以具有数字原始声音的样本为单位合成关于测量部分的信息，并输出合成结果。 此外，提供了一种将参考声音的测量部分与可变带宽多编解码器的端对端音质测量进行匹配的方法以及由变量产生的声音的测量部分 带宽多编解码器在实时互联网多媒体服务。 因此，可以减少由于不匹配的测量部分引起的测量结果的失真。

公开(公告)号：US07986634B2

公开(公告)日：2011-07-26

申请号：US11857539

申请日：2007-09-19

申请人： Tae-Gyu Kang ,  Ki-Jong Koo ,  Dae-Ho Kim ,  Do Young Kim ,  Hae Won Jung

发明人： Tae-Gyu Kang ,  Ki-Jong Koo ,  Dae-Ho Kim ,  Do Young Kim ,  Hae Won Jung

IPC分类号： H04L12/26

CPC分类号： G10L25/69 ,  G10L19/24

摘要： Provided are a method and apparatus for measuring sound quality in a variable band multi-codec. The sound quality measurement apparatus includes: a recording file receiving/generating unit receiving a first recording file in which a natural sound is recorded, and a second recording file obtained by converting the natural sound into digital data using the variable band multi-codec, receiving information obtained by encoding the natural sound using the variable band multi-codec, in the format of a Real Time Protocol (RTP) packet, unpacking the RTP packet, decoding the RTP packet using the variable band multi-codec, and generating a third recording file; a Mean Opinion Score (MOS) value calculating unit repeatedly selecting a file from among the first recording file, the second recording file, and the third recording file, or selecting two files from among the first recording file, the second recording file, and the third recording file, and calculating a MOS value by obtaining a difference between the selected results; and a MOS value comparison unit comparing a plurality of MOS values generated by the MOS value calculating unit, with each other, and detecting a cause of sound quality deterioration.

摘要翻译： 提供了一种用于测量可变频带多编解码器中的声音质量的方法和装置。 声音测量装置包括：记录文件接收/生​​成单元，接收其中记录有自然声音的第一记录文件;以及第二记录文件，其通过使用可变频带多编解码器将自然声音转换为数字数据而获得，第二记录文件接收 使用可变频带多编解码器以实时协议（RTP）分组的格式编码自然声音获得的信息，解包RTP分组，使用可变频带多编解码器解码RTP分组，以及生成第三记录 文件; 平均意见得分（MOS）值计算单元，重复从第一记录文件，第二记录文件和第三记录文件中选择文件，或者从第一记录文件，第二记录文件和第二记录文件中选择两个文件 第三记录文件，并且通过获得所选择的结果之间的差来计算MOS值; 以及MOS值比较单元，将由MOS值计算单元生成的多个MOS值彼此进行比较，并且检测出音质劣化的原因。

公开(公告)号：US07778177B2

公开(公告)日：2010-08-17

申请号：US11634561

申请日：2006-12-06

申请人： Tae Gyu Kang ,  Do Young Kim ,  Hae Won Jung

发明人： Tae Gyu Kang ,  Do Young Kim ,  Hae Won Jung

IPC分类号： H04J1/16

CPC分类号： H04W28/18

摘要： An apparatus and method of variable bandwidth multi-codec quality of service (QoS) control are provided. The apparatus for controlling the QoS of a variable bandwidth multi-codec includes: a network state detection unit detecting a network state including at least one of a packet loss ratio, a packet loss interval, and a packet delay time based on an RTP packet transmitted to and received from a destination for which a call connection is established; and a codec control unit updating a transmission rate by comparing the detected resultant value with an already set reference value and increasing or decreasing the transmission rate, and controlling the variable bandwidth multi-codec to code data with the updated transmission rate. According to the apparatus and method, data can be coded with a codec transmission rate suitable for a network state identified during a voice call after the call is set up.

摘要翻译： 提供了一种可变带宽多编解码器服务质量（QoS）控制的装置和方法。 用于控制可变带宽多编解码器的QoS的装置包括：网络状态检测单元，其基于发送的RTP分组检测包括分组丢失率，分组丢失间隔和分组延迟时间中的至少一个的网络状态 从建立呼叫连接的目的地接收和接收; 以及编解码器控制单元，通过将检测到的结果值与已经设定的基准值进行比较，并且增加或减少传输速率来更新传输速率，并且控制可变带宽多编解码器以更新的传输速率对数据进行编码。 根据该装置和方法，可以在呼叫建立之后利用适合于在语音呼叫期间识别的网络状态的编解码传输速率对数据进行编码。

公开(公告)号：US20070133441A1

公开(公告)日：2007-06-14

申请号：US11634561

申请日：2006-12-06

申请人： Tae Gyu Kang ,  Do Young Kim ,  Hae Won Jung

发明人： Tae Gyu Kang ,  Do Young Kim ,  Hae Won Jung

IPC分类号： H04B1/44

CPC分类号： H04W28/18

摘要： An apparatus and method of variable bandwidth multi-codec quality of service (QoS) control are provided. The apparatus for controlling the QoS of a variable bandwidth multi-codec includes: a network state detection unit detecting a network state including at least one of a packet loss ratio, a packet loss interval, and a packet delay time based on an RTP packet transmitted to and received from a destination for which a call connection is established; and a codec control unit updating a transmission rate by comparing the detected resultant value with an already set reference value and increasing or decreasing the transmission rate, and controlling the variable bandwidth multi-codec to code data with the updated transmission rate. According to the apparatus and method, data can be coded with a codec transmission rate suitable for a network state identified during a voice call after the call is set up.

摘要翻译： 提供了一种可变带宽多编解码器服务质量（QoS）控制的装置和方法。 用于控制可变带宽多编解码器的QoS的装置包括：网络状态检测单元，其基于发送的RTP分组检测包括分组丢失率，分组丢失间隔和分组延迟时间中的至少一个的网络状态 从建立呼叫连接的目的地接收和接收; 以及编解码器控制单元，通过将检测到的结果值与已经设定的参考值进行比较来增加或减少传输速率来更新传输速率，并且控制可变带宽多编解码器以更新的传输速率对数据进行编码。 根据该装置和方法，可以在呼叫建立之后利用适合于在语音呼叫期间识别的网络状态的编解码传输速率对数据进行编码。

公开(公告)号：US08958296B2

公开(公告)日：2015-02-17

申请号：US13112189

申请日：2011-05-20

申请人： Ki Jong Koo ,  Do Young Kim ,  Byung Sun Lee

发明人： Ki Jong Koo ,  Do Young Kim ,  Byung Sun Lee

IPC分类号： H04W48/20 ,  H04W48/12 ,  H04L12/26

CPC分类号： H04W48/12 ,  H04L43/0876 ,  H04W48/20

摘要： An access point (AP) measures a congestion level of a transmission channel representing a collision probability between frames at a channel busy duration and transmits the congestion level to a terminal. Therefore, the terminal extracts a congestion level that is included in a frame that is received from the AP through a scan process for searching for an AP to which the terminal is to connect. The terminal selects an AP to connect from at least one AP based on the extracted congestion level of the transmission channel.

摘要翻译： 接入点（AP）测量表示在频道忙时间的帧之间的冲突概率的传输信道的拥塞级别，并将拥塞级别发送到终端。 因此，终端通过用于搜索终端要连接的AP的扫描处理来提取包含在从AP接收的帧中的拥塞级别。 终端基于提取的传输信道的拥塞级别，选择AP至少从一个AP进行连接。

公开(公告)号：US20110286329A1

公开(公告)日：2011-11-24

申请号：US13112189

申请日：2011-05-20

申请人： Ki Jong Koo ,  Do Young Kim ,  Byung Sun Lee

发明人： Ki Jong Koo ,  Do Young Kim ,  Byung Sun Lee

IPC分类号： H04W24/00 ,  H04W84/02 ,  H04L12/26

CPC分类号： H04W48/12 ,  H04L43/0876 ,  H04W48/20

摘要： An access point (AP) measures a congestion level of a transmission channel representing a collision probability between frames at a channel busy duration and transmits the congestion level to a terminal. Therefore, the terminal extracts a congestion level that is included in a frame that is received from the AP through a scan process for searching for an AP to which the terminal is to connect. The terminal selects an AP to connect from at least one AP based on the extracted congestion level of the transmission channel.

摘要翻译： 接入点（AP）测量表示在频道忙时间的帧之间的冲突概率的传输信道的拥塞级别，并将拥塞级别发送到终端。 因此，终端通过用于搜索终端要连接的AP的扫描处理来提取包含在从AP接收的帧中的拥塞级别。 终端基于提取的传输信道的拥塞级别，选择AP至少从一个AP进行连接。

公开(公告)号：US07477647B2

公开(公告)日：2009-01-13

申请号：US10837292

申请日：2004-04-29

申请人： Seunghan Choi ,  Tae-Gyu Kang ,  Do Young Kim

发明人： Seunghan Choi ,  Tae-Gyu Kang ,  Do Young Kim

IPC分类号： H04L12/28

CPC分类号： H04L65/104 ,  H04L29/06027 ,  H04L65/103 ,  H04L65/1043 ,  H04L69/08 ,  H04M7/1255 ,  H04M7/126

摘要： The present invention relates to a method of controlling common call connection and a media gateway for executing the method, which are capable of performing call connection control without a change in a control procedure under different protocols and the interworking of the different protocols. In the method of the present invention, each of a plurality of termination points connected by a call in the media gateway, each of a plurality of multimedia services transmitted and received between the terminations through connection setup of the call and each of a plurality of physical resources per multimedia service in the multimedia are defined as a termination, a stream and a channel, respectively, a call processing request based on one of various media gateway control protocols is mapped into a model defined as described above, and resource allocation, connection setup or release with respect to a channel belonging to a call and a termination are requested according to the type of the call processing request. Accordingly, the method of the present invention accommodates call processing based on various types of protocols.

摘要翻译： 本发明涉及一种控制公共呼叫连接的方法和一种用于执行该方法的媒体网关，其能够在不同协议下的控制过程中进行呼叫连接控制而不进行调整，以及不同协议的互通。 在本发明的方法中，通过媒体网关中的呼叫连接的多个终端中的每一个，通过呼叫的连接建立和多个物理的每个之间在终端之间发送和接收的多个多媒体服务中的每一个 将多媒体中的每个多媒体业务的资源分别定义为终端，流和信道，基于各种媒体网关控制协议之一的呼叫处理请求被映射到如上所述定义的模型中，资源分配，连接建立 或根据呼叫处理请求的类型请求关于属于呼叫的信道和终止的释放。 因此，本发明的方法适用于基于各种协议的呼叫处理。