DTMF tone signal transmission method and DTMF tone signal transmission system
    1.
    发明申请
    DTMF tone signal transmission method and DTMF tone signal transmission system 有权
    DTMF音信号传输方式和DTMF音信号传输系统

    公开(公告)号:US20050243872A1

    公开(公告)日:2005-11-03

    申请号:US11104484

    申请日:2005-04-13

    申请人: Nobuhiro Monai

    发明人: Nobuhiro Monai

    摘要: It is made possible for a DTMF tone signal associated with a button depressed on an IP telephone terminal to arrive at the opposite party even if an IP telephone terminal does not have a function of sending a DTMF tone signal. If a DTMF tone signal transmission request signal detection unit in a proxy detects a DTMF tone signal transmission request signal inserted in an RTP packet or a SIP packet received from an IP telephone terminal or detects a DTMF digit signal contained in the DTMF tone signal transmission request signal, a DTMF tone signal generation unit generates a DTMF tone signal associated with the DTMF digit signal. A voice encoder encodes the DTMF tone signal in the same way as a voice signal. An RTP payload replacement unit replaces a voice signal in a payload of the RTP packet with the encoded DTMF tone signal.

    摘要翻译: 即使IP电话终端不具有发送DTMF音信号的功能,也可以使与IP电话终端上按下的按钮相关联的DTMF音信号到达对方。 如果代理中的DTMF音信号发送请求信号检测单元检测到插入到RTP分组中的DTMF音信号发送请求信号或从IP电话终端接收到的SIP分组,或者检测DTMF音信号发送请求中包含的DTMF数字信号 信号,DTMF音信号产生单元产生与DTMF数字信号相关联的DTMF音信号。 语音编码器以与语音信号相同的方式对DTMF音信号进行编码。 RTP有效载荷替换单元用编码的DTMF音调信号替换RTP分组的有效载荷中的语音信号。

    System for automatically selecting voice data transmission and reception system for IP network, method thereof, and IP terminal
    2.
    发明申请
    System for automatically selecting voice data transmission and reception system for IP network, method thereof, and IP terminal 有权
    用于自动选择IP网络语音数据收发系统,方法和IP终端的系统

    公开(公告)号:US20050099996A1

    公开(公告)日:2005-05-12

    申请号:US10654128

    申请日:2003-09-04

    摘要: Voice is input to a voice input and output portion 11. Thereafter, a digital-analog portion converts analog signal of voice into digital data. An RTP port confirming portion confirms whether or not a communication according to the RTP can be made. When the communication according to the RTP cannot be made, with reference to a history table, a port through which a voice data communication can be made is determined. Digitized voice is sent to a call controlling portion. The call controlling portion performs a call control for the voice communication. When the communication according to the RTP can be made, the voice data is sent to an IP communication portion. The IP communication portion paketizes the voice data. Packetized voice data according to the IP is sent to a LAN interface portion. The LAN interface portion outputs the packetized voice data to a network.

    摘要翻译: 语音被输入到语音输入和输出部分11。 此后,数字模拟部分将语音的模拟信号转换为数字数据。 RTP端口确认部分确认是否可以进行根据RTP的通信。 当不能进行根据RTP的通信时,参考历史表,确定能够进行语音数据通信的端口。 数字化语音被发送到呼叫控制部分。 呼叫控制部分执行语音通信的呼叫控制。 当可以进行根据RTP的通信时,语音数据被发送到IP通信部分。 IP通信部分分配语音数据。 根据IP的分组语音数据被发送到LAN接口部分。 LAN接口部分将分组语音数据输出到网络。

    IP telephony method and IP telephone system
    3.
    发明申请
    IP telephony method and IP telephone system 有权
    IP电话方式和IP电话系统

    公开(公告)号:US20050207431A1

    公开(公告)日:2005-09-22

    申请号:US11078341

    申请日:2005-03-14

    申请人: Nobuhiro Monai

    发明人: Nobuhiro Monai

    摘要: An IP telephony method facilitating transferring an IP telephony communication inside a LAN is realized. A telephone conversation takes place between a first IP telephone terminal and a conversation partner's IP telephone terminal through a router and a proxy. The proxy relays the telephone conversation by referring to a proxy table that holds a correspondence among a global IP address and a receive port of the conversation partner's IP telephone terminal, and a local IP address and a receive port of the first IP telephone terminal during relay. To perform transfer from the first IP telephone terminal to a second IP telephone terminal, the local IP address and the receive port of the first IP telephone terminal in the proxy table are changed to a local IP address and a receive port of the second IP telephone terminal, respectively.

    摘要翻译: 实现了便于在LAN内部传送IP电话通信的IP电话方法。 通过路由器和代理在第一IP电话终端和对话伙伴的IP电话终端之间进行电话通话。 代理通过参考保持对话伙伴的IP电话终端的全局IP地址和接收端口之间的对应关系的代理表以及中继期间第一IP电话终端的本地IP地址和接收端口来中继电话会话 。 为了执行从第一IP电话终端到第二IP电话终端的传送,代理表中的第一IP电话终端的本地IP地址和接收端口被改变为第二IP电话的本地IP地址和接收端口 终端。

    IP telephony method and IP telephone system
    4.
    发明授权
    IP telephony method and IP telephone system 有权
    IP电话方式和IP电话系统

    公开(公告)号:US07508818B2

    公开(公告)日:2009-03-24

    申请号:US11078341

    申请日:2005-03-14

    申请人: Nobuhiro Monai

    发明人: Nobuhiro Monai

    IPC分类号: H04L12/66 H04L12/28

    摘要: An IP telephony method facilitating transferring an IP telephony communication inside a LAN is realized. A telephone conversation takes place between a first IP telephone terminal and a conversation partner's IP telephone terminal through a router and a proxy. The proxy relays the telephone conversation by referring to a proxy table that holds a correspondence among a global IP address and a receive port of the conversation partner's IP telephone terminal, and a local IP address and a receive port of the first IP telephone terminal during relay. To perform transfer from the first IP telephone terminal to a second IP telephone terminal, the local IP address and the receive port of the first IP telephone terminal in the proxy table are changed to a local IP address and a receive port of the second IP telephone terminal, respectively.

    摘要翻译: 实现了便于在LAN内部传送IP电话通信的IP电话方法。 通过路由器和代理在第一IP电话终端和对话伙伴的IP电话终端之间进行电话通话。 代理通过参考保持对话伙伴的IP电话终端的全局IP地址和接收端口之间的对应关系的代理表以及中继期间第一IP电话终端的本地IP地址和接收端口来中继电话会话 。 为了执行从第一IP电话终端到第二IP电话终端的传送,代理表中的第一IP电话终端的本地IP地址和接收端口被改变为第二IP电话的本地IP地址和接收端口 终端。

    DTMF tone signal transmission method and DTMF tone signal transmission system
    5.
    发明授权
    DTMF tone signal transmission method and DTMF tone signal transmission system 有权
    DTMF音信号传输方式和DTMF音信号传输系统

    公开(公告)号:US08311033B2

    公开(公告)日:2012-11-13

    申请号:US11104484

    申请日:2005-04-13

    申请人: Nobuhiro Monai

    发明人: Nobuhiro Monai

    IPC分类号: H04L12/66 H04J3/12

    摘要: It is made possible for a DTMF tone signal associated with a button depressed on an IP telephone terminal to arrive at the opposite party even if an IP telephone terminal does not have a function of sending a DTMF tone signal. If a DTMF tone signal transmission request signal detection unit in a proxy detects a DTMF tone signal transmission request signal inserted in an RTP packet or a SIP packet received from an IP telephone terminal or detects a DTMF digit signal contained in the DTMF tone signal transmission request signal, a DTMF tone signal generation unit generates a DTMF tone signal associated with the DTMF digit signal. A voice encoder encodes the DTMF tone signal in the same way as a voice signal. An RTP payload replacement unit replaces a voice signal in a payload of the RTP packet with the encoded DTMF tone signal.

    摘要翻译: 即使IP电话终端不具有发送DTMF音信号的功能,也可以使与IP电话终端上按下的按钮相关联的DTMF音信号到达对方。 如果代理中的DTMF音信号发送请求信号检测单元检测到插入到RTP分组中的DTMF音信号发送请求信号或从IP电话终端接收到的SIP分组,或者检测DTMF音信号发送请求中包含的DTMF数字信号 信号,DTMF音信号产生单元产生与DTMF数字信号相关联的DTMF音信号。 语音编码器以与语音信号相同的方式对DTMF音信号进行编码。 RTP有效载荷替换单元用编码的DTMF音调信号替换RTP分组的有效载荷中的语音信号。

    System for automatically selecting voice data transmission and reception system for IP network, method thereof, and IP terminal
    6.
    发明授权
    System for automatically selecting voice data transmission and reception system for IP network, method thereof, and IP terminal 有权
    用于自动选择IP网络语音数据收发系统,方法和IP终端的系统

    公开(公告)号:US07301937B2

    公开(公告)日:2007-11-27

    申请号:US10654128

    申请日:2003-09-04

    IPC分类号: H04L12/66

    摘要: Voice is input to a voice input and output portion 11. Thereafter, a digital-analog portion converts analog signal of voice into digital data. An RTP port confirming portion confirms whether or not a communication according to the RTP can be made. When the communication according to the RTP cannot be made, with reference to a history table, a port through which a voice data communication can be made is determined. Digitized voice is sent to a call controlling portion. The call controlling portion performs a call control for the voice communication. When the communication according to the RTP can be made, the voice data is sent to an IP communication portion. The IP communication portion paketizes the voice data. Packetized voice data according to the IP is sent to a LAN interface portion. The LAN interface portion outputs the packetized voice data to a network.

    摘要翻译: 语音被输入到语音输入和输出部分11。 此后,数字模拟部分将语音的模拟信号转换为数字数据。 RTP端口确认部分确认是否可以进行根据RTP的通信。 当不能进行根据RTP的通信时,参考历史表,确定能够进行语音数据通信的端口。 数字化语音被发送到呼叫控制部分。 呼叫控制部分执行语音通信的呼叫控制。 当可以进行根据RTP的通信时,语音数据被发送到IP通信部分。 IP通信部分分配语音数据。 根据IP的分组语音数据被发送到LAN接口部分。 LAN接口部分将分组语音数据输出到网络。