WO2009097764A1 - 一种实现语音呼叫的方法、系统及装置 - Google Patents

一种实现语音呼叫的方法、系统及装置 Download PDF

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Publication number
WO2009097764A1
WO2009097764A1 PCT/CN2009/070121 CN2009070121W WO2009097764A1 WO 2009097764 A1 WO2009097764 A1 WO 2009097764A1 CN 2009070121 W CN2009070121 W CN 2009070121W WO 2009097764 A1 WO2009097764 A1 WO 2009097764A1
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WIPO (PCT)
Prior art keywords
call
information
voice
computer
voice call
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PCT/CN2009/070121
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English (en)
French (fr)
Inventor
Kai Zhang
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Shenzhen Huawei Communication Technologies Co., Ltd.
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Application filed by Shenzhen Huawei Communication Technologies Co., Ltd. filed Critical Shenzhen Huawei Communication Technologies Co., Ltd.
Priority to EP09708763A priority Critical patent/EP2120423A4/en
Publication of WO2009097764A1 publication Critical patent/WO2009097764A1/zh

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/253Telephone sets using digital voice transmission
    • H04M1/2535Telephone sets using digital voice transmission adapted for voice communication over an Internet Protocol [IP] network
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/247Telephone sets including user guidance or feature selection means facilitating their use
    • H04M1/2473Telephone terminals interfacing a personal computer, e.g. using an API (Application Programming Interface)

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a system, apparatus, and method for implementing a voice call.
  • VoIP Voice over Internet Protocol
  • the current mainstream communication method is to use traditional telephone network (PSTN) terminals (such as ordinary telephones) +
  • PSTN telephone network
  • the VoIP voice processing terminal combines the way to perform call operations, and this VoIP call has a tendency to replace the traditional P STN voice call.
  • PC existing computer
  • the call of the PC that is, the communication parties need to be both PCs, and the two parties must be online at the same time to realize the actual VoIP communication.
  • such applications cannot directly implement VoIP communication between PCs and PSTN terminals under the constraints and control of existing telecommunications open policies and other factors.
  • the inventors have also found in the process of implementing the present invention that some early products have passed the conversion of narrowband analog signaling.
  • the called PSTN terminal directly responds.
  • a narrow-band calling card to be installed on the PC, which is inconvenient to apply.
  • Embodiments of the present invention provide a method, system, and device for implementing a voice call, which can implement a voice call by using a computer, and use a voice network terminal to perform VoIP voice communication, which is simple, fast, convenient, and low-cost.
  • a voice call is initiated to the called user, and the current call state information of the voice call is fed back to the computer and the telephone network terminal, respectively.
  • a system for implementing a voice call includes a computer, a VoIP terminal, and a telephone network terminal, where:
  • a computer configured to receive call information determined by the calling user, after performing voice call processing on the call information, sending the call information to a VoIP terminal, receiving the current call state information, and the current call state Information feedback to the calling user;
  • a VoIP terminal configured to receive call information sent by the computer, initiate a voice call to the called user according to the call information, and feed back current call state information of the voice call to the computer and the telephone network terminal respectively;
  • a telephone network terminal configured to receive the current call state information, feed back the current call state information to a calling user, and in the case that the calling user selects a telephone network terminal to perform VoIP voice communication, and the VoIP The terminal performs information exchange and completes the voice call.
  • the device for implementing a voice call is loaded into a computer, and the device includes: a called user information providing module, configured to provide called user information to the calling user through an interactive interface; and a call information receiving module, configured to receive Call information determined by the calling user;
  • the voice call processing module is configured to perform voice call processing on the call information received by the call information receiving module, and send the processed call information to the VoIP terminal.
  • the VoIP terminal for implementing a voice call includes: [24] a call information receiving processing module, configured to receive call information sent by a computer, and perform internal protocol processing on the received call information;
  • a voice call initiating module configured to send a voice call to the called user according to the call information processed by the internal protocol
  • Call status information feedback module used to feed the current call status information to the computer and the telephone network terminal respectively.
  • a voice call is initiated to a called user according to the call information sent by the received computer, and the current call state information is separately fed back to the computer.
  • a telephone network terminal thereby realizing a Vol P voice call simply, quickly, conveniently, and at low cost.
  • the VoIP terminal can feed the call status information to the computer to which the calling user belongs and the telephone network terminal, so that the calling user can perform voice communication according to personal habits.
  • FIG. 1 is a schematic structural diagram of the system according to an embodiment of the present disclosure
  • FIG. 2 is a schematic structural diagram of implementing a voice call apparatus in the PC according to an embodiment of the present invention
  • FIG. 3 is a schematic structural diagram of the VoIP terminal according to an embodiment of the present invention.
  • FIG. 4 is a schematic diagram of a specific implementation process of the method according to an embodiment of the present invention.
  • FIG. 1 A system for implementing a voice call according to an embodiment of the present invention, as shown in FIG. 1, specifically by a computer (P)
  • the PC1 10 involved in the embodiment of the present invention may include, in addition to the functional modules of the ordinary PC, a voice call device 120, which may be a software program and stored in any one of the saveable devices.
  • the medium of the software program can be loaded by any device that can load the software program. It can be understood that the device 120 can also be composed of hardware to implement a voice call function.
  • the implementation of the voice call device 120, as shown in FIG. 2, may specifically include: [39] 1)
  • the called user information providing module 121 is configured to provide the calling user through the interactive interface. Called user
  • the called user information providing module 121 can be understood as a module having a phone book function in the PC 110, and is used to store all called user information related to the calling user, including the called user name, number, and address. Information such as age, gender, etc., and the above information can be entered and changed by the calling user.
  • the called user information providing module 121 may further have a searching function. When the calling user inputs any one or more pieces of information of the called user information, the called user information providing module 121 may correspondingly provide the related information related to the input information. Call the user all the information.
  • the called user information providing module 121 can be activated through an interactive interface (usually the interactive interface is a display), and the called user is determined by inputting or searching. interest.
  • the calling user can also input the information of the called user directly through the interactive interface, thereby initiating a voice call.
  • the call information receiving module 122 is configured to receive call information determined by the calling user.
  • the call information receiving module 122 receives the called user information determined by the calling user, that is, the call information. And sending the call information to the voice call processing module 123.
  • the voice call processing module 123 is configured to perform voice call processing on the call information received by the call information receiving module 122, and send the processed call information to the VoIP terminal 130.
  • the voice call processing performed by the voice call processing module 123 on the call information can be understood as the voice dialing signaling constructed based on the call information determined by the calling user. For example, using the Session Initiation Protocol (SIP)
  • SIP Session Initiation Protocol
  • SIP client SIP client proxy
  • the embodiment of the present invention is not limited to the use of SIP as the communication protocol, and the embodiment of the present invention can be applied to any practical application by using any communication protocol.
  • the call status information feedback module 124 is configured to feed back the call status information fed back by the VoIP terminal 130 to the calling user through the interaction interface.
  • the VoIP terminal 130 can feed back the current call state information to the PC 110 and the telephone network terminal 140.
  • the call state information feedback module 124 in the PC 110 can feed back the current call state information fed back by the VoIP terminal 130 to the calling user through the interactive interface, so as to facilitate the user to perform corresponding processing.
  • the above current call status information is called, the other party is busy, the other party picks up the phone, and the like.
  • the call status information feedback module 124 can feed back the current call status information to the calling user in the form of animation, video, sound, picture, and the like.
  • the voice call device 120 further provided by the embodiment of the present invention may further include:
  • the communication module 125 is configured to complete voice communication by interacting with the Vo IP terminal 130 when the calling user selects the PC 110 to perform VoIP voice communication.
  • the communication module 125 performs signaling and data message transmission by interacting with the VoIP terminal 130, and passes through external devices of the PC 110, such as a headset, a microphone, and the like.
  • the received voice information of the calling user is transmitted to the called user, and the voice information of the called user is transmitted to the calling user to complete the VoIP voice communication between the calling party and the called party.
  • the PC 110 can establish a communication connection with the VoIP terminal 130 through a standard Ethernet interface, a USB interface, a wireless interface, and the like.
  • the VoIP terminal 130 is configured to receive call information sent by the PC 110, and initiate a voice call to the called user according to the call information.
  • the VoIP terminal 130 can also be used to feed back current call state information to the PC 110 and the telephony network terminal 140, respectively.
  • the VoIP terminal 130 may further include:
  • the call information receiving processing module 131 is configured to receive call information sent by the PC 110, and perform internal protocol processing on the received call information.
  • the VoIP terminal 130 When the VoIP terminal 130 receives the call information sent by the PC 110, the VoIP terminal 130 performs internal protocol processing on the received call information, such as the back-to-back user agent (B2BUA) processing function of the SIP protocol, and records the completion of the internal protocol processing. Call information.
  • B2BUA back-to-back user agent
  • the call information receiving processing module 131 is also used to terminate the voice call processing of the PC 110.
  • the voice call initiation module 132 is configured to initiate a voice call to the called user according to the call information processed by the internal protocol.
  • the voice call initiation module 132 re-sends a voice call to the called user according to the call information processed by the internal protocol.
  • the call status information feedback module 133 is configured to feed back the current call status information to the PC 110 and the telephone network terminal 140, respectively.
  • the VoIP terminal 130 can also send current call state information to the PC 110 and the telephony network terminal 140. Specifically, the VoIP terminal 130 performs the copying operation on the received signaling of the communication terminal to which the called user belongs, such as the current call state information such as the calling party, the other party busy, and the other party's off-hook, and respectively feeds back the copied current state information. To PC 110 and telephone network terminal 140.
  • the process of copying the current call state information by the VoIP terminal 130 may specifically be performed by extending the B2BUA processing function in the SIP protocol to have a copy function, and copying the signaling message representing the current call state information, and The copied signaling messages are fed back to the PC 110 and the telephone network terminal 140, respectively.
  • the P C110 and the telephone network terminal 140 perform corresponding responses according to the received signaling message, such as playing a ring back tone. Further, the user can select the PC 110 or the telephone network terminal 140 to answer and complete the voice communication according to the usage habit.
  • Vo IP terminal 130 has the function of copying the current call state, and may be based on the PC 110 and the telephone network connected in the system.
  • the number of terminals 140 determines the number of copies of current call state information.
  • the VoIP terminal 130 performs corresponding operations according to the current call state, which may specifically include the following situations:
  • the VoIP terminal 130 constructs call signaling information for the telephone network terminal 140, and plays a ring back tone for the telephone network terminal 140;
  • the VoIP terminal 130 establishes a call connection between the calling party and the called party.
  • the operation of the PC 110 after receiving the call state information fed back by the VoIP terminal 130 has already been mentioned in the above description of the PC 110 call state information feedback module 124, and therefore will not be described.
  • the telephone network terminal 140 is configured to perform information interaction with the VoIP terminal 130 to complete a voice call.
  • the telephone network terminal 140 involved in the embodiment of the present invention may be any type of telephone network terminal, such as a general telephone. Its function is basically the same as that of a normal telephone.
  • the telephone network terminal 140 can be connected to the VoIP terminal 130 via a twisted pair.
  • function modules included in the implementation of the voice call system provided by the embodiments of the present invention can be implemented by software code and stored in any storage medium.
  • the system for implementing a voice call adds a corresponding function module to the PC 110 and the VoIP terminal 130, and through the human-machine interaction, after the call information determined by the calling user is processed by the voice call,
  • the VoIP terminal 130 initiates a voice call to the called user according to the call information, thereby implementing a VoIP voice call simply, quickly, and conveniently.
  • the PC 110 has been widely used, it is advantageous for the wide application of the voice call system provided by the embodiment of the present invention.
  • the voice call system provided by the embodiment of the present invention does not need to modify the existing device, and only needs to load the software application.
  • the program can be used, therefore, it has the advantages of convenient operation, quickness, simplicity, and reliability.
  • the voice call system provided by the embodiment of the present invention can be respectively fed back to the PC 110 and the telephone network terminal 140 of the calling user after the current call state information is copied, so the user can select the PC 110 or the ordinary according to the usage habit. Voice communication is performed on the telephone 140.
  • the embodiment of the present invention further provides a method for implementing a voice call, by receiving call information sent by the PC 110, according to the received call information, initiating a voice call to the called user, and feeding back the current call state to the call state respectively.
  • the PC 110 and the telephone network terminal 140 enable VoIP voice calls to be implemented simply, quickly, conveniently, and at low cost.
  • the voice call method provided by the embodiment of the present invention is applied to the voice call system provided in the embodiment of the present invention, and the voice call provided by the embodiment of the present invention is implemented in conjunction with FIG.
  • the specific implementation process of the method is described in detail. The process may specifically include:
  • Step 41 the calling user initiates a call operation.
  • the called user information providing module 121 in the PC 110 can be activated by any means to input or find the information of the called user to determine the call information.
  • Step 42 The PC 110 performs voice call processing on the call information determined by the calling user, and sends the processed call information to the VoIP terminal 130.
  • the voice call processing module 123 in the Betsey PC1 10 performs voice call processing on the call information, and The processed call information is sent to the VoIP terminal 130.
  • Step 43 After receiving the call information, the VoIP terminal 130 performs internal protocol processing on the call information, and initiates a voice call to the called user according to the call information processed by the internal protocol.
  • the VoIP terminal 130 When the VoIP terminal 130 receives the call information sent by the PC1 10, the VoIP terminal 130 performs internal protocol processing on the received call information, for example, B2BUA processing using the SIP protocol, and records the call information after the internal protocol is processed, and then A voice call is initiated to the called user according to the call information processed by the internal protocol.
  • internal protocol processing for example, B2BUA processing using the SIP protocol
  • Step 44 the VoIP terminal 130 feeds back the current call state information.
  • the VoIP terminal feeds back the current call state information fed back by the called user's terminal to the PC1 10 and the telephone network terminal 140, respectively.
  • the VoIP terminal 130 may specifically extend a certain processing function of a certain communication protocol to have a function of copying a signaling message representing the current call state information, and according to the PC1 10 and the telephone network terminal 140 in the system.
  • the number of the signaling messages of the current call state information is copied to the corresponding number and fed back to the PC 1 10 and the telephone network terminal 140 respectively.
  • Step 45 The PC1 10 and the telephone network terminal 140 perform corresponding operations according to the current call state information fed back by the VoIP terminal 130.
  • the PC1 10 when the PC1 10 receives the current call state information fed back by the VoIP terminal 130, such as a call, a busy party, an off-hook, etc., the PC 1 10 can present the current form through animation, video, sound, picture, and the like.
  • the call status information is fed back to the calling user through the interactive interface.
  • the calling user picks up the telephone network terminal 140, and thereafter, the VoIP terminal 130 can perceive that the telephone network terminal 140 has gone off-hook, and therefore, the VoIP terminal 130 establishes with the telephone network terminal 140.
  • the communication contact feeds the current call status information to the telephone network terminal 140. If the current called user does not go off-hook, the VoIP terminal 130 constructs signaling for playing the ring back tone, and the calling user can receive the call by the telephone network terminal 140.
  • the ringback tone if the currently called user picks up the phone, the VoIP terminal 130 establishes a communication connection between the telephone network terminal 140 to which the calling user belongs and the communication terminal to which the called user belongs.
  • Step 46 the calling user selects PC1 10 or telephone network terminal 140 for voice communication.
  • the calling user can select PC 110 or telephone network terminal 140 to perform voice communication with the called user according to his own usage habits.
  • the method for implementing a voice call adds a corresponding function module to the PC 110 and the VoIP terminal 130, and passes the human-machine interaction to pass the call information determined by the calling user.
  • the VoIP terminal 130 initiates a voice call to the called user according to the call information, thereby implementing a VoIP voice call simply, quickly, conveniently, and at low cost.
  • the PC 110 has been widely used, it is advantageous for the wide application of the method for implementing the voice call provided by the embodiment of the present invention.
  • the method for implementing the voice call provided by the embodiment of the present invention does not need to modify the existing device, and only needs to load the software application.
  • the program can be used, therefore, it has the advantages of convenient operation, quickness, simplicity, and reliability.
  • the method for implementing a voice call provided by the embodiment of the present invention can copy the current call state information to the PC 110 and the telephone network terminal 140 of the calling user, so that the user can select the PC 110 or the ordinary phone according to the usage habit.
  • the machine 140 performs voice communication.
  • the method, system, and apparatus for implementing a voice call provided by the embodiment of the present invention are also applicable to a voice called process, that is, when a called communication service occurs in a communication terminal to which the user belongs, VoIP
  • the terminal 130 may perform the copy operation on the received called information, and send the copied called information to the PC 110 and the telephone network terminal 140 to which the user belongs, and the PC 110 and the telephone network terminal 140 may perform the call according to the received called information.
  • Corresponding responses, such as ringing, etc. the user can select PC 110 or telephone network terminal 140 to answer according to personal usage habits. Therefore, the above technical solutions should also be within the scope of protection of the embodiments of the present invention.

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Telephonic Communication Services (AREA)
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Abstract

本发明涉及一种实现语音呼叫的方法、系统及装置。本发明实施例中,通过在计算机以及VoIP终端中增加相应的功能模块,并通过人机交互,在将主叫用户确定的呼叫信息经过语音呼叫处理后,由VoIP终端根据所述呼叫信息向被叫用户发起语音呼叫,从而简单、快捷、方便的实现VoIP语音呼叫;同时,本发明实施例可以将此次语音呼叫的当前呼叫状态信息反馈至主叫用户所属计算机和电话网络终端,使用户可以在电话网络终端或计算机继续本次呼叫,这样可以不影响用户的使用习惯,方便用户在例如普通电话机上进行通话。

Description

一种实现语音呼叫的方法、 系统及装置
[1] 本申请要求于 2008年 2月 4日提交中国专利局、 申请号为 200810057689.6、 发明 名称为"一种实现语音呼叫的方法、 系统及装置"的中国专利申请的优先权, 其全 部内容通过引用结合在本申请中。
[2] 技术领域
[3] 本发明涉及通信技术领域, 尤其涉及一种实现语音呼叫的系统、 装置及方法。
[4] 发明背景
互联网语音处理技术 (VoIP) 是指在使用了互联网协议的网络上进行语音传输 , 由于釆用 VoIP进行语音通信可以降低通信资费, 因此, VoIP已经成为使用最 为广泛的通信产品之一。
[6] 目前主流的通信方式还是釆用传统电话网络 (PSTN) 终端 (比如普通话机) +
VoIP话音处理终端结合的方式来进行呼叫操作, 这种 VoIP呼叫已经有取代传统 P STN语音呼叫的趋势。
[7] 发明人在实现本发明的过程中发现, 现有技术情况下, 用户在传统电话网络终 端上实现 VoIP呼叫, 需要先找到电话本的电话号码, 再拨号进行呼叫。 由于普 通 PSTN线路不支持或者不能方便、 有效的提供在线査号功能进行拨打, 因此, 这种方式在拨叫过程中需要反复按键, 如果按键错误、 或者呼叫未接通、 或者 对方占线, 用户还须再次重复上述操作, 过程比较繁琐。 虽然有些传统电话网 络终端可以存储电话号码, 或者提供屏幕进行交互査找, 但此类传统电话网络 终端通常成本较高, 额外增加了用户的资费负担。 另外, 利用传统电话网络终 端进行呼叫操作还存在操作不方便, 文字录入、 査找易用性不好等问题。
发明人在实现本发明的过程中还发现, 现有计算机 (PC) 中的软件也可以实现 VoIP呼叫, 地址本査找也很方便, 但此类应用实现的是 PC to
PC的呼叫, 即需要通信双方都是 PC机, 通信双方必须同吋在线才可能实现实吋 的 VoIP通信。 而且, 此类应用在现有电信开放政策等因素的制约和控制下, 无 法实现 PC机与 PSTN终端之间直接进行 VoIP通信。 发明人在实现本发明的过程中还发现, 有些早期产品通过窄带模拟信令的转换
, 或者通过并机来达到 PC发起呼叫, 被叫 PSTN终端直接应答。 但这样的系统, 需要在 PC安装一个窄带电话卡, 应用不方便。
发明内容
本发明实施例提供一种实现语音呼叫的方法、 系统及装置, 可以简单、 快捷、 方便、 低成本的实现釆用计算机发起语音呼叫, 釆用电话网络终端等实体进行 V oIP语音通信。
本发明实施例提供的实现语音呼叫的方法, 包括:
接收计算机发送的呼叫信息;
向被叫用户发起语音呼叫, 并将该语音呼叫的当前呼叫状态信息分别反馈至计 算机和电话网络终端。
本发明实施例提供的实现语音呼叫的系统, 包括计算机、 VoIP终端和电话网络 终端, 其中:
计算机, 用于接收主叫用户确定的呼叫信息, 在对所述呼叫信息进行语音呼叫 处理后, 将所述呼叫信息发送至 VoIP终端, 接收所述当前呼叫状态信息, 并将 所述当前呼叫状态信息反馈至主叫用户;
VoIP终端, 用于接收所述计算机发送的呼叫信息, 根据所述呼叫信息向被叫用 户发起语音呼叫, 并将所述语音呼叫的当前呼叫状态信息分别反馈至计算机和 电话网络终端;
电话网络终端, 用于接收所述当前呼叫状态信息, 将所述当前呼叫状态信息反 馈至主叫用户, 并在所述主叫用户选择电话网络终端进行 VoIP语音通信的情况 下, 与所述 VoIP终端进行信息交互, 完成语音呼叫。
本发明实施例提供的实现语音呼叫的装置加载于计算机中, 该装置包括: 被叫用户信息提供模块, 用于通过交互界面为主叫用户提供被叫用户信息; 呼叫信息接收模块, 用于接收主叫用户确定的呼叫信息;
语音呼叫处理模块, 用于对所述呼叫信息接收模块接收的呼叫信息进行语音呼 叫处理, 并将处理后的呼叫信息发送至 VoIP终端。
本发明实施例提供的实现语音呼叫的 VoIP终端, 包括: [24] 呼叫信息接收处理模块, 用于接收计算机发送的呼叫信息, 并对接收的所述呼 叫信息进行内部协议处理;
[25] 语音呼叫发起模块, 用于根据所述经过内部协议处理的呼叫信息向被叫用户发 起语音呼叫;
[26] 呼叫状态信息反馈模块, 用于将当前呼叫状态信息分别反馈至计算机和电话网 络终端。
[27] 由上述本发明实施例提供的技术方案可以看出, 本发明实施例中, 通过根据接 收的计算机发送的呼叫信息向被叫用户发起语音呼叫, 并将当前呼叫状态信息 分别反馈至计算机和电话网络终端, 从而简单、 快捷、 方便、 低成本的实现 Vol P语音呼叫。 同吋, VoIP终端能够将呼叫状态信息反馈至主叫用户所属计算机以 及电话网络终端, 方便主叫用户根据个人习惯进行语音通信。
[28] 附图简要说明
[29] 图 1为本发明实施例提供的所述系统结构示意图;
[30] 图 2为本发明实施例提供的所述 PC机内实现语音呼叫装置结构示意图;
[31] 图 3位本发明实施例提供的所述 VoIP终端结构示意图;
[32] 图 4为本发明实施例提供的所述方法具体实现过程示意图。
[33] 实施本发明的方式
[34] 为了便于理解, 下面结合附图对本发明实施例进行详细表述。
[35] 本发明实施例提供的实现语音呼叫的系统, 如附图 1所示, 具体可由计算机 (P
C) 1 10、 互联网语音处理技术 (VoIP) 终端 130以及电话网络 (PSTN) 终端 140 组成。 其中:
[36] 1、 PC1 10 , 用于为主叫用户提供被叫用户信息, 并接收主叫用户确定的呼叫信 息, 在对呼叫信息进行语音呼叫处理后, 将完成语音呼叫处理的呼叫信息发送 至 VoIP终端 130。
[37] 本发明实施例中所涉及的 PC1 10 , 除了包括普通 PC具有的功能模块外, 具体还 可以包括实现语音呼叫装置 120, 该装置 120可为软件程序, 并存储于任一种可 保存软件程序的媒介中, 可由任一种可加载软件程序的设备加载。 可以理解的 是, 该装置 120也可由硬件组成, 实现语音呼叫功能。 [38] 本发明实施例提供的实现语音呼叫装置 120, 如附图 2所示, 具体可以包括: [39] 1) 被叫用户信息提供模块 121, 用于通过交互界面, 为主叫用户提供被叫用户
Ι π Λ∑!、。
[40] 具体的, 被叫用户信息提供模块 121可以理解为 PC110中具有电话记录本功能的 模块, 用于储存与主叫用户有关的所有被叫用户信息, 包括被叫用户名称、 号 码、 地址、 年龄、 性别等信息, 而且, 上述信息可以由主叫用户随吋输入及更 改。 被叫用户信息提供模块 121还可以具有査找功能, 当主叫用户输入上述被叫 用户信息中的任一项或多项信息, 被叫用户信息提供模块 121会相应的提供与输 入信息相关的被叫用户所有信息。
[41] 当主叫用户需要发起语音呼叫吋, 可通过交互界面 (通常情况下交互界面即为 显示器) 激活被叫用户信息提供模块 121, 并通过输入或査找等方式, 确定被叫 用户 ί曰息。
[42] 可以理解的是, 主叫用户也可以直接通过交互界面, 输入被叫用户的信息, 从 而发起语音呼叫。
[43] 2) 呼叫信息接收模块 122, 用于接收主叫用户确定的呼叫信息。
[44] 当主叫用户输入或査找到被叫用户信息后, 可通过点击确定键或者双击等方式 确定, 此吋, 呼叫信息接收模块 122接收主叫用户确定的被叫用户信息, 即呼叫 信息, 并将呼叫信息发送至语音呼叫处理模块 123。
[45] 3) 语音呼叫处理模块 123, 用于对呼叫信息接收模块 122接收的呼叫信息进行 语音呼叫处理, 并将处理后的呼叫信息发送至 VoIP终端 130。
[46] 语音呼叫处理模块 123对呼叫信息进行的语音呼叫处理, 其实质可以理解为根 据主叫用户确定的呼叫信息, 构造语音拨号信令。 比如釆用会话发起协议 (SIP
) , 作为 SIP客户端代理 (SIP client
UA) , 发起对被叫用户的邀请 (Invite) 消息。
[47] 需要说明的是, 本发明实施例在具体实现过程中, 并不仅限于釆用 SIP作为通 信协议, 本发明实施例可釆用任一种通信协议进行实际应用。
[48] 当语音呼叫处理模块 123完成对呼叫信息的语音呼叫处理后, 将呼叫信息发送 至系统内的 VoIP终端 130。 [49] 4) 呼叫状态信息反馈模块 124, 用于通过交互界面将 VoIP终端 130反馈的呼叫 状态信息反馈至主叫用户。
[50] 具体的, 当 VoIP终端 130根据呼叫信息发起向被叫用户的语音呼叫吋, VoIP终 端 130可以将当前的呼叫状态信息反馈至 PC110以及电话网络终端 140。 此吋, PC 110中的呼叫状态信息反馈模块 124可以将 VoIP终端 130反馈的当前呼叫状态信息 通过交互界面反馈至主叫用户, 以方便用户进行相应的处理。 上述当前呼叫状 态信息如正在呼叫、 对方忙、 对方摘机等。 呼叫状态信息反馈模块 124可以以动 画、 影像、 声音、 图片等形式向主叫用户反馈当前呼叫状态信息。
[51] 本发明实施例提供的实现语音呼叫装置 120进一步还可以包括:
[52] 5) 通信模块 125, 用于当主叫用户选择 PC110进行 VoIP语音通信吋, 通过与 Vo IP终端 130交互, 完成语音通信。
[53] 当主叫用户选择 PC110进行 VoIP语音通信吋, 所述通信模块 125通过与 VoIP终 端 130交互, 完成信令和数据报文的传输, 并通过 PC110的外部设备, 比如耳机 、 话筒等, 将接收的主叫用户的语音信息传输至被叫用户, 以及将被叫用户的 语音信息传输至主叫用户, 完成主、 被叫用户之间的 VoIP语音通信。
[54] PC110可通过标准以太网接口、 USB接口、 无线接口等接口与 VoIP终端 130建 立通信连接。
[55] 2、 VoIP终端 130, 用于接收 PC110发送的呼叫信息, 并根据呼叫信息向被叫用 户发起语音呼叫。
[56] VoIP终端 130还可以用于将当前的呼叫状态信息分别反馈至 PC110和电话网络 终端 140。
[57] 如附图 3所示,
VoIP终端 130除具备普通 VoIP终端的功能模块外, 具体还可以包括:
[58] 1) 呼叫信息接收处理模块 131, 用于接收 PC110发送的呼叫信息, 并对接收的 呼叫信息进行内部协议处理。
[59] 当 VoIP终端 130接收到 PC110发送的呼叫信息, VoIP终端 130将接收的呼叫信息 进行内部协议处理, 比如釆用 SIP协议的背靠背用户代理 (B2BUA) 处理功能, 并记录完成内部协议处理后的呼叫信息。 [60] 呼叫信息接收处理模块 131还用于终结 PC110的语音呼叫处理。
[61] 2) 语音呼叫发起模块 132, 用于根据经过内部协议处理的呼叫信息向被叫用户 发起语音呼叫。
[62] 语音呼叫发起模块 132根据完成内部协议处理后的呼叫信息, 重新向被叫用户 发起语音呼叫。
[63] 3) 呼叫状态信息反馈模块 133, 用于将当前呼叫状态信息分别反馈至 PC110和 电话网络终端 140。
[64] VoIP终端 130还可以将当前的呼叫状态信息发送至 PC110以及电话网络终端 140 。 具体的, VoIP终端 130将接收的被叫用户所属通信终端回复的信令, 比如正在 呼叫、 对方忙、 对方摘机等当前呼叫状态信息, 进行复制操作, 并将复制后的 当前状态信息分别反馈至 PC110以及电话网络终端 140。
[65] VoIP终端 130对当前呼叫状态信息进行复制的过程, 具体可以通过扩展 SIP协议 中的 B2BUA处理功能, 使其具有复制的功能, 将代表当前呼叫状态信息的信令 报文进行复制, 并将复制后的信令报文分别反馈至 PC110和电话网络终端 140。 P C110和电话网络终端 140根据接收的信令报文进行相应的响应, 比如播放回铃音 等。 进而, 用户可根据使用习惯, 选择釆用 PC110或者电话网络终端 140进行接 听, 并完成语音通信。
[66] 可以理解的是, 本发明实施例中, 也可以扩展其他通信协议的某个功能, 使 Vo IP终端 130具有复制当前呼叫状态的功能, 并且, 可以根据系统中连接的 PC110 和电话网络终端 140的数量决定复制当前呼叫状态信息的个数。
[67] 举例说明, 若在语音呼叫过程中, 与 VoIP终端 130相连的电话网络终端 140摘机 , 则 VoIP终端 130根据当前呼叫状态进行相应的操作, 具体可以包括以下情况:
[68] 若此吋被叫用户所属终端尚未应答, 处于回铃音状态, 则 VoIP终端 130为电话 网络终端 140构建呼叫信令信息, 为电话网络终端 140播放回铃音;
[69] 若此吋被叫用户所属终端摘机, 则 VoIP终端 130建立主、 被叫之间的通话连接
[70] PC110在接收到 VoIP终端 130反馈的呼叫状态信息后的操作, 在上述对 PC110呼 叫状态信息反馈模块 124的描述过程中已经提及, 因此不再赞述。 [71] 3) 电话网络终端 140, 用于与 VoIP终端 130进行信息交互, 完成语音呼叫。
[72] 本发明实施例中所涉及的电话网络终端 140, 可以为任一种电话网络终端, 如 普通电话机等。 其作用与普通电话机基本相同。
[73] 电话网络终端 140可通过双绞线与 VoIP终端 130相连。
[74] 可以理解的是, 上述本发明实施例提供的实现语音呼叫系统中所包括的功能模 块, 可以由软件代码实现其功能, 并保存在任意存储介质内。
[75] 本发明实施例提供的实现语音呼叫的系统, 通过在 PC110以及 VoIP终端 130内 增加相应的功能模块, 并通过人机交互, 在将主叫用户确定的呼叫信息经过语 音呼叫处理后, 由 VoIP终端 130根据呼叫信息向被叫用户发起语音呼叫, 从而简 单、 快捷、 方便的实现 VoIP语音呼叫。 由于 PC110已经广泛普及, 因此, 有利于 本发明实施例提供的实现语音呼叫系统的广泛应用, 而且, 本发明实施例提供 的实现语音呼叫系统, 无需对现有设备进行改造, 只需加载软件应用程序即可 , 因此, 具有操作方便、 快捷、 简单、 可靠等优点。 同吋, 本发明实施例提供 的实现语音呼叫系统, 能够在将当前呼叫状态信息进行复制后, 分别反馈至主 叫用户的 PC110和电话网络终端 140, 所以用户可以根据使用习惯, 选择 PC110或 普通电话机 140上进行语音通信。
[76] 本发明实施例还提供了一种实现语音呼叫的方法, 通过接收 PC110发送的呼叫 信息, 根据接收的所述呼叫信息, 向被叫用户发起语音呼叫, 并将当前呼叫状 态分别反馈至 PC110和电话网络终端 140, 从而简单、 快捷、 方便、 低成本的实 现 VoIP语音呼叫。
[77] 为了便于理解, 下面以本发明实施例提供的实现语音呼叫方法, 应用于本发明 实施例提供的实现语音呼叫系统为例, 并结合附图 4, 对本发明实施例提供的实 现语音呼叫方法的具体实现过程进行详细的描述, 所述过程具体可以包括:
[78] 步骤 41, 主叫用户发起呼叫操作。
[79] 当主叫用户需要进行语音呼叫吋, 可以通过任一种方式激活 PC110中的被叫用 户信息提供模块 121, 输入或査找被叫用户的信息, 确定呼叫信息。
[80] 步骤 42, PC110对主叫用户确定的呼叫信息进行语音呼叫处理, 并将处理后的 呼叫信息发送至 VoIP终端 130。 [81] 当主叫用户确定被叫用户信息 (通常情况下是被叫用户所属终端号码) , 即呼 叫信息, 贝岫 PC1 10中的语音呼叫处理模块 123对呼叫信息进行语音呼叫处理, 并将处理后的呼叫信息发送至 VoIP终端 130。
[82] 步骤 43, VoIP终端 130在接收到呼叫信息后, 对其进行内部协议处理, 并根据 内部协议处理后的呼叫信息向被叫用户发起语音呼叫。
[83] 当 VoIP终端 130接收到 PC1 10发送的呼叫信息, VoIP终端 130将接收的呼叫信息 进行内部协议处理, 比如釆用 SIP协议的 B2BUA处理, 并记录完成内部协议处理 后的呼叫信息, 然后根据完成内部协议处理后的呼叫信息, 向被叫用户发起语 音呼叫。
[84] 步骤 44, VoIP终端 130反馈当前呼叫状态信息。
[85] VoIP终端将被叫用户所属终端反馈的当前呼叫状态信息, 分别反馈至 PC1 10和 电话网络终端 140。
[86] VoIP终端 130具体可以通过扩展某个通信协议的某个处理功能, 使其具有复制 代表当前呼叫状态信息的信令报文的功能, 并根据系统中 PC1 10和电话网络终端 140的个数, 将当前呼叫状态信息的信令报文复制相应的个数, 并分别反馈至 PC 1 10和电话网络终端处 140。
[87] 步骤 45 , PC1 10和电话网络终端 140根据 VoIP终端 130反馈的当前呼叫状态信息 , 进行相应的操作。
[88] 具体的, 当 PC1 10接收到 VoIP终端 130反馈的当前呼叫状态信息, 比如正在呼 叫、 对方忙、 对方摘机等吋, PC1 10可以通过动画、 影像、 声音、 图片等表现形 式将当前呼叫状态信息通过交互界面反馈至主叫用户。
[89] 当呼叫过程中, 主叫用户将电话网络终端 140摘机, 此吋, VoIP终端 130能够感 知到电话网络终端 140已经摘机, 因此, VoIP终端 130建立与电话网络终端 140之 间的通信联系, 将当前呼叫状态信息反馈至电话网络终端 140, 若当前被叫用户 没有摘机, 则 VoIP终端 130构建播放回铃音的信令, 主叫用户可以通过电话网络 终端 140接听到正在呼叫的回铃音, 若当前被叫用户摘机, 则 VoIP终端 130建立 主叫用户所属电话网络终端 140与被叫用户所属通信终端之间的通信连接。
[90] 步骤 46, 主叫用户选择 PC1 10或电话网络终端 140进行语音通信。 [91] 主叫用户可以根据自己的使用习惯, 选择 PC110或者电话网络终端 140, 与被叫 用户进行语音通信。
[92] 通过上述描述可知, 本发明实施例提供的实现语音呼叫的方法, 通过在 PC110 以及 VoIP终端 130内增加相应的功能模块, 并通过人机交互, 在将主叫用户确定 的呼叫信息经过语音呼叫处理后, 由 VoIP终端 130根据呼叫信息向被叫用户发起 语音呼叫, 从而简单、 快捷、 方便、 低成本的实现 VoIP语音呼叫。 由于 PC110已 经广泛普及, 因此, 有利于本发明实施例提供的实现语音呼叫方法的广泛应用 , 而且, 本发明实施例提供的实现语音呼叫方法, 无需对现有设备进行改造, 只需加载软件应用程序即可, 因此, 具有操作方便、 快捷、 简单、 可靠等优点 。 同吋, 本发明实施例提供的实现语音呼叫方法, 能够将当前呼叫状态信息进 行复制后, 分别反馈至主叫用户的 PC110和电话网络终端 140, 所以用户可以根 据使用习惯, 选择 PC110或普通电话机 140进行语音通信。
[93] 进一步说明的是, 本发明实施例提供的实现语音呼叫的方法、 系统及装置, 也 同样可以适用于语音被叫的过程中, 即当用户所属通信终端发生被叫通信业务 吋, VoIP终端 130可以将接收的被叫信息进行复制操作, 并将复制后的被叫信息 分别发送至用户所属 PC110和电话网络终端 140, PC 110和电话网络终端 140可以 根据接收到的被叫信息, 进行相应的响应, 比如响铃等, 用户可以根据个人的 使用习惯, 选择 PC110或电话网络终端 140进行接听。 因此, 上述技术方案也应 该在本发明实施例保护的范围之内。
[94] 以上所述, 仅为本发明较佳的具体实施方式, 但本发明的保护范围并不局限于 此, 任何熟悉本技术领域的技术人员在本发明揭露的技术范围内, 可轻易想到 的变化或替换, 都应涵盖在本发明的保护范围之内。 因此, 本发明的保护范围 应该以权利要求的保护范围为准。

Claims

权利要求书
[1] 一种实现语音呼叫的方法, 其特征在于, 包括:
接收计算机发送的呼叫信息;
向被叫用户发起语音呼叫, 并将该语音呼叫的当前呼叫状态信息分别反馈 至计算机和电话网络终端。
[2] 根据权利要求 1所述的方法, 其特征在于, 所述呼叫信息为:
所述计算机对主叫用户确定的呼叫信息进行语音呼叫处理后的呼叫信息。
[3] 根据权利要求 1或 2所述的方法, 其特征在于, 所述向被叫用户发起语音呼 叫的过程包括:
对接收的所述呼叫信息进行内部协议处理;
记录所述完成内部协议处理的呼叫信息, 并根据所述完成内部协议处理的 呼叫信息, 向被叫用户发起语音呼叫。
[4] 根据权利要求 1所述的方法, 其特征在于, 在所述将当前呼叫状态信息分别 反馈至计算机和电话网络终端之后, 所述方法还包括:
所述计算机和电话网络终端接收所述当前呼叫状态信息, 并将所述当前呼 叫状态信息反馈至主叫用户。
[5] 根据权利要求 1所述的方法, 其特征在于, 所述方法还包括:
接收主叫方发送来的被叫信息, 并将所述被叫信息分别反馈至所述计算机 和电话网络终端。
[6] 一种实现语音呼叫的系统, 其特征在于, 包括计算机、 VoIP终端和电话网 络终端, 其中:
计算机, 用于接收主叫用户确定的呼叫信息, 在对所述呼叫信息进行语音 呼叫处理后, 将所述呼叫信息发送至 VoIP终端, 接收所述当前呼叫状态信 息, 并将所述当前呼叫状态信息反馈至主叫用户;
VoIP终端, 用于接收所述计算机发送的呼叫信息, 根据所述呼叫信息向被 叫用户发起语音呼叫, 并将所述语音呼叫的当前呼叫状态信息分别反馈至 计算机和电话网络终端;
电话网络终端, 用于接收所述当前呼叫状态信息, 将所述当前呼叫状态信 息反馈至主叫用户, 并在所述主叫用户选择电话网络终端进行 VoIP语音通 信的情况下, 与所述 VoIP终端进行信息交互, 完成语音呼叫。
[7] 根据权利要求 6所述的系统, 其特征在于, 所述计算机通过标准以太网接口
、 或者 USB接口、 或者无线接口, 与所述 VoIP终端连接。
[8] —种实现语音呼叫的装置, 其特征在于, 所述装置加载于计算机中, 包括 被叫用户信息提供模块, 用于通过交互界面为主叫用户提供被叫用户信息 呼叫信息接收模块, 用于接收主叫用户确定的呼叫信息;
语音呼叫处理模块, 用于对所述呼叫信息接收模块接收的呼叫信息进行语 音呼叫处理, 并将处理后的呼叫信息发送至 VoIP终端。
[9] 根据权利要求 8所述的装置, 其特征在于, 所述装置还包括:
呼叫状态信息反馈模块, 用于通过交互界面将 VoIP终端反馈的当前呼叫状 态信息反馈至主叫用户;
通信模块, 用于当主叫用户选择计算机进行 VoIP语音通信吋, 通过与 VoIP 终端交互, 完成语音通信。
[10] 一种实现语音呼叫的 VoIP终端, 其特征在于, 包括:
呼叫信息接收处理模块, 用于接收计算机发送的呼叫信息, 并对接收的所 述呼叫信息进行内部协议处理;
语音呼叫发起模块, 用于根据所述经过内部协议处理的呼叫信息向被叫用 户发起语音呼叫;
呼叫状态信息反馈模块, 用于将当前呼叫状态信息分别反馈至计算机和电 话网络终端。
[11] 根据权利要求 10所述的 VoIP终端, 其特征在于, 所述呼叫信息接收处理模 块还用于终结计算机的语音呼叫处理。
PCT/CN2009/070121 2008-02-04 2009-01-13 一种实现语音呼叫的方法、系统及装置 WO2009097764A1 (zh)

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