TWI552568B - Initially initiated by the establishment of a two-way call - Google Patents
Initially initiated by the establishment of a two-way call Download PDFInfo
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- TWI552568B TWI552568B TW100125733A TW100125733A TWI552568B TW I552568 B TWI552568 B TW I552568B TW 100125733 A TW100125733 A TW 100125733A TW 100125733 A TW100125733 A TW 100125733A TW I552568 B TWI552568 B TW I552568B
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Description
本發明係關於一種由外部發起建立雙方通話之方法,特別為一種應用於NGN/IMS網路並使用SIP通信協定來建立發話方與受話方兩端通話之方法。 The present invention relates to a method for establishing a call between two parties by external means, in particular, a method for applying to an NGN/IMS network and using a SIP communication protocol to establish a call between a caller and a callee.
由外部發起建立雙方通話之方法,過去多為透過網頁點擊方式來建立雙方通話。先前技術受限於傳統固定或行動交換網路,都是採用伺服器橋接方式,先以伺服器對發話方發話,建立伺服器與發話方通話,然後再由伺服器對受話方發話,建立伺服器與受話方之通話,最後伺服器再以橋接的方式將發話方與受話方之語音內容交互傳遞來達到雙方通話之目的。 The method of establishing a call between two parties is initiated by the outside. In the past, the call between the two parties was established through a webpage click. The prior art is limited to the traditional fixed or mobile switching network. The server bridges the server. The server first talks to the sender, establishes the server to talk with the caller, and then the server sends a call to the caller to establish the servo. The call between the device and the called party, and finally the server relays the voice content of the Talker and the Receiver in a bridging manner to achieve the purpose of the two parties.
目前習知的由外部發起建立雙方通話方法中,台灣專利申請案(I324010)結合網頁之即時雙向通話連結方法,係以伺服器對發話方發話,建立伺服器與發話方通話,由伺服器對受話方發話,建立伺服器與受話方之通話,再接續發話方與受話方兩端通話,發話方與受話方雙方需透過伺服器橋接兩端通話。前案是透過伺服器橋接發話方與受話方兩端通話,因此,發話方與受話方間所有的聲音或影像都必須透過伺服器傳送,對伺服器負載要求極高。且當發話方與受話方兩端通話時,若伺服器發生障礙則會造成發話方與受話方之通話中斷。 In the current method of establishing a two-party call by externally, the Taiwan patent application (I324010) combines the instant two-way call connection method of the webpage, and the server sends a call to the utterer to establish a call between the server and the utterer, and the server pairs The called party sends a call to establish a call between the server and the called party, and then the caller and the callee both end the call, and both the caller and the callee need to bridge the two ends through the server. The former case is to bridge the caller and the caller through the server. Therefore, all the voices or images between the caller and the caller must be transmitted through the server, which requires extremely high load on the server. And when the caller and the called party talk at both ends, if the server is in trouble, the call between the sender and the caller will be interrupted.
由於前案是透過伺服器橋接發話方與受話方兩端通話,因此,發話方與受話方間所有的語音內容都必須透過伺服器進行交互傳送,對 伺服器負載要求極高,且當發話方與受話方兩端通話時,若伺服器發生障礙則會造成發話方與受話方之通話中斷。若發話方與受話方為非VoIP(Voice over IP)電話而是一般傳統電話,伺服器必須申請門號電路才能對發話方及受話方發話。若發話方與受話方雙方皆有語音信箱,則可能會造成發話方與受話方雙方語音信箱間通話之問題。 Since the previous case is to bridge the Talker and the called party through the server, all the voice content between the Talker and the Receiver must be transmitted through the server. The server load requirement is extremely high, and when the caller and the callee bother to talk, if the server is in trouble, the call between the sender and the caller will be interrupted. If the calling party and the receiving party are non-VoIP (Voice over IP) phones but are generally traditional phones, the server must apply for the door number circuit to speak to the calling party and the called party. If both the caller and the callee have voicemail, it may cause a problem between the voicemail of the caller and the callee.
有鑑於以上種種,發明人經努力研究與實驗,設計出本發明之一種由外部發起建立雙方通話之方法。本發明之技術是利用NGN/IMS之SIP信號去建立雙方通話,通話建立完成後,發話方與受話方雙方之通話並不透過伺服器橋接。因此,伺服器障礙不影響發話方與受話方既有之通話狀態。且伺服器本身為NGN/IMS網路上一節點,不需要任何電路。另外,本發明之建立雙方通話之方法中運用多媒體伺服器對發話端進行請求按碼確認,可避免雙方語音信箱通話之問題。 In view of the above, the inventors have worked hard to study and experiment, and have devised a method for establishing a call between two parties by the outside. The technology of the present invention utilizes the SIP signal of NGN/IMS to establish a two-party call. After the call is established, the call between the caller and the callee is not bridged through the server. Therefore, the server barrier does not affect the conversation state between the Talker and the recipient. And the server itself is a node on the NGN/IMS network and does not require any circuitry. In addition, in the method for establishing a two-party call in the present invention, a multimedia server is used to perform a request for confirmation by the source terminal, and the problem of voicemail conversation between the two parties can be avoided.
本發明之目的,在提供一種由外部發起建立雙方通話之方法,提供外部資訊應用系統一個傳輸控制協議/互聯網絡協議(Transmission Control Protocol/Internet Protocol,以下簡稱TCP/IP)網路介面,並使用SIP通信協定於NGN/IMS網路上,來幫助外部資訊系統建立發話方與受話方兩端通話,並解決因語音信箱而造成雙方語音信箱通話之問題。 The purpose of the present invention is to provide a method for establishing a call between two parties by externally, and to provide a transmission control protocol/Internet Protocol (TCP/IP) network interface of the external information application system, and use The SIP communication protocol is used on the NGN/IMS network to help the external information system establish a call between the caller and the callee, and solve the problem of voicemail conversation between the two parties due to voicemail.
達成上述發明目的之由外部發起建立雙方通話之方法,其系統使用模組架構區分為(一)流程控制單元,執行、控制、管理服務流程,接受外部資訊系統之請求,並整合各子單元以完成服務;(二)身分及 邏輯驗證單元,負責進行外部資訊系統之身分驗證,以及發話方與受話方之號碼驗證;(三)訊號控制單元,用以界接NGN/IMS網路,進行SIP信號之收送與編解碼。並搭配多媒體伺服器來達成由外部發起建立雙方通話之服務。 The method for establishing a call between two parties is achieved by externally, and the system uses a module architecture to distinguish (1) a process control unit, execute, control, and manage a service flow, accept a request from an external information system, and integrate each subunit. Complete the service; (2) identity and The logic verification unit is responsible for performing identity verification of the external information system and verifying the number of the calling party and the called party. (3) The signal control unit is used to interface the NGN/IMS network to perform SIP signal transmission and encoding and decoding. And with a multimedia server to achieve the service initiated by the outside to establish a two-party call.
本發明之方法係利用流程控制單元來接受外部資訊系統所發起之建立雙方通話之要求。經由身分與邏輯驗證單元對外部資訊系統之身分驗證,以及發話方與受話方之號碼驗證。通過驗證後,流程控制單元透過信號控制單元以SIP信號於NGN/IMS網路上發出發話方與多媒體伺服器建立通話之請求。先將發話端接續至多媒體伺服,由多媒體伺服器播放要求發話方按鍵之語音,待發話方按下按鍵後,多媒體伺服器則切斷與發話方之通話。此時信號控制單元接受到多媒體伺服器切斷通話信號後告知流程控制單元。流程控制單元再透過信號控制單元對NGN/IMS網路送出將發話方與受話方建立連線之請求來建立雙方通話。 The method of the present invention utilizes a flow control unit to accept a request initiated by an external information system to establish a two-party call. The identity verification of the external information system by the identity and logic verification unit, and the verification of the number of the utterer and the callee. After verification, the flow control unit sends a request for a call between the Talker and the multimedia server to the NGN/IMS network via the signal control unit via the SIP signal. The Talker is connected to the multimedia servo first, and the multimedia server plays the voice of the Talker button. When the Talker presses the button, the multimedia server cuts off the call with the Talker. At this time, the signal control unit informs the flow control unit after receiving the multimedia server to cut off the call signal. The flow control unit then sends a request to the NGN/IMS network to establish a connection between the Talker and the called party through the signal control unit to establish a two-party call.
請參閱圖一所示,為本發明之系統架構示意圖,本發明包含一流程控制單元61,負責執行、控制、管理服務流程,接受外部資訊系統之請求,並整合各子單元以完成服務;一身分及邏輯驗證單元62,負責進行外部資訊系統之身分驗證,以及發話方與受話方之號碼驗證;一訊號控制單元63,用以界接NGN/IMS網路,進行SIP信號之收送與編解碼。 Please refer to FIG. 1 , which is a schematic diagram of a system architecture of the present invention. The present invention includes a process control unit 61 that is responsible for executing, controlling, and managing service processes, accepting requests from external information systems, and integrating sub-units to complete services; The identity and logic verification unit 62 is responsible for performing identity verification of the external information system and verifying the number of the caller and the callee. A signal control unit 63 is used to interface the NGN/IMS network to perform SIP signal transmission and editing. decoding.
請參閱圖二所示,為本發明之方法流程圖,其步驟包含:a.外部 資訊系統透過TCP/IP網路,對本發明系統之流程控制單元發出建立雙方通話之請求;b.流程控制單元透過身分及邏輯驗證單元驗證資訊系統之身分,並驗證發話方與受話方之電話號碼;c.流程控制單元在驗證通過後,要求信號控制單元發出建立多媒體伺服器與發話方通話之要求;d.信號控制單元透過NGN/IMS網路以SIP通信協定對多媒體伺服器發出建立與發話方通話之請求;e.信號控制單元透過NGN/IMS網路以SIP通信協定對發話方發出建立與多媒體伺服器通話之請求;f.多媒體伺服器與發話方通話建立成功後,要求發話方進行按鍵動作,來確定發話方非自然人以外之語音系統或設備;g.發話方按鍵;h.多媒體伺服器切斷與發話方之通話後,信號控制單元由NGN/IMS網路接受SIP通信協定多媒體伺服器斷線訊息,並通知流程控制單元;i.流程控制單元要求信號控制單元透過NGN/IMS網路以SIP通信協定對發話方發出建立與受話方通話之請求;j.流程控制單元要求信號控制單元透過NGN/IMS網路以SIP通信協定對受話方發出建立與發話方通話之請求;k.發話方與受話方通話。 Please refer to FIG. 2, which is a flowchart of the method of the present invention, and the steps thereof include: a. external The information system sends a request for establishing a two-party call to the process control unit of the system of the present invention through the TCP/IP network; b. the process control unit verifies the identity of the information system through the identity and logic verification unit, and verifies the telephone number of the caller and the callee After the verification is passed, the process control unit requests the signal control unit to issue a request to establish a multimedia server to talk to the caller; d. the signal control unit sends and establishes a voice message to the multimedia server through the NGN/IMS network in a SIP communication protocol. The request of the party call; e. The signal control unit sends a request to establish a call with the multimedia server to the Talker through the NGN/IMS network in the SIP communication protocol; f. After the multimedia server and the Talker successfully establish a call, the Talker is required to perform the call. Key action to determine the voice system or device other than the non-natural person; g. Talker button; h. After the multimedia server cuts off the call with the Talker, the signal control unit accepts the SIP communication protocol multimedia by the NGN/IMS network The server disconnects the message and notifies the flow control unit; i. the flow control unit requests the signal control unit to pass The NGN/IMS network sends a request to establish a call with the caller to the caller in the SIP communication protocol; j. The flow control unit requests the signal control unit to issue a call with the caller to the caller via the NGN/IMS network in the SIP communication protocol. Request; k. The caller speaks with the callee.
本發明係提供外部資訊應用系統一個TCP/IP網路介面,並使用SIP通信協定於NGN/IMS網路上,來幫助外部資訊系統建立發話方與受話方兩端通話之服務。其中外部資訊系統可以是任何可透過TCP/IP網路傳遞資訊的設備,包含網頁伺服器(例如網站部落格、拍賣網站、企業公司首頁)、電腦資訊系統(例如企業員工通訊錄系統)、電腦應用程式、隨選式多媒體電視(Multimedia On Demand,MOD)/網路電視(IPTV)系統…等。另外發話方與受話方可為一般第二代行動通訊/第三代行動 通訊(2/3G)行動電話、室內電話、NGN網路電話皆可。 The present invention provides a TCP/IP network interface for an external information application system and uses a SIP communication protocol on the NGN/IMS network to help the external information system establish a service for both the caller and the callee to talk at both ends. The external information system can be any device that can transmit information through the TCP/IP network, including a web server (such as a website blog, an auction website, a corporate company home page), a computer information system (such as a corporate employee address book system), and a computer. Applications, Multimedia On Demand (MOD) / Internet TV (IPTV) systems, etc. In addition, the Talker and the Receiver can be the general second generation mobile communication / third generation action Communication (2/3G) mobile phones, indoor phones, and NGN Internet phones are all available.
10‧‧‧NGN/IMS網路 10‧‧‧NGN/IMS network
20‧‧‧TCP/IP網路 20‧‧‧TCP/IP network
30‧‧‧外部資訊系統 30‧‧‧External Information System
40‧‧‧發話方 40‧‧‧Talker
50‧‧‧受話方 50‧‧‧Recipients
60‧‧‧本系統伺服器 60‧‧‧System server
61‧‧‧流程控制單元 61‧‧‧Process Control Unit
62‧‧‧身分及邏輯驗證單元 62‧‧‧Identity and logic verification unit
63‧‧‧信號控制單元 63‧‧‧Signal Control Unit
70‧‧‧多媒體伺服器 70‧‧‧Multimedia server
請參閱以下有關本發明之詳細說明及其附圖,將可進一步瞭解本發明之技術內容及其目的功效;有關附圖為:圖一為本發明之系統架構示意圖;圖二為本發明之方法流程圖; The following is a detailed description of the present invention and its accompanying drawings, and the technical contents of the present invention and the functions thereof can be further understood; the related drawings are: FIG. 1 is a schematic diagram of the system architecture of the present invention; FIG. 2 is a method of the present invention. flow chart;
10‧‧‧NGN/IMS網路 10‧‧‧NGN/IMS network
20‧‧‧TCP/IP網路 20‧‧‧TCP/IP network
30‧‧‧外部資訊系統 30‧‧‧External Information System
40‧‧‧發話方 40‧‧‧Talker
50‧‧‧受話方 50‧‧‧Recipients
60‧‧‧本系統伺服器 60‧‧‧System server
61‧‧‧流程控制單元 61‧‧‧Process Control Unit
62‧‧‧身分及邏輯驗證單元 62‧‧‧Identity and logic verification unit
63‧‧‧信號控制單元 63‧‧‧Signal Control Unit
70‧‧‧多媒體伺服器 70‧‧‧Multimedia server
Claims (8)
Priority Applications (2)
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TW100125733A TWI552568B (en) | 2011-07-21 | 2011-07-21 | Initially initiated by the establishment of a two-way call |
CN201110339367.2A CN102333102B (en) | 2011-07-21 | 2011-11-01 | System and method for establishing double-party call initiated from outside |
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TW100125733A TWI552568B (en) | 2011-07-21 | 2011-07-21 | Initially initiated by the establishment of a two-way call |
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TW201306552A TW201306552A (en) | 2013-02-01 |
TWI552568B true TWI552568B (en) | 2016-10-01 |
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TW100125733A TWI552568B (en) | 2011-07-21 | 2011-07-21 | Initially initiated by the establishment of a two-way call |
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TW (1) | TWI552568B (en) |
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TW201406125A (en) * | 2012-07-20 | 2014-02-01 | Nen-Fu Huang | Telephone communication method and system thereof |
TWI563820B (en) * | 2013-01-10 | 2016-12-21 | Nen Fu Huang | Telephone communication method and system thereof |
CN105430207A (en) * | 2015-11-11 | 2016-03-23 | 北京易车互联信息技术有限公司 | Two-way communication control method and system |
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CN101335794A (en) * | 2007-06-27 | 2008-12-31 | 田种楠 | Network telephone connecting system and network telephone connecting method |
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- 2011-07-21 TW TW100125733A patent/TWI552568B/en not_active IP Right Cessation
- 2011-11-01 CN CN201110339367.2A patent/CN102333102B/en not_active Expired - Fee Related
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US5818836A (en) * | 1995-08-09 | 1998-10-06 | Duval; Stephen C. | Method and apparatus for anonymous voice communication using an online data service |
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CN102333102A (en) | 2012-01-25 |
CN102333102B (en) | 2014-06-18 |
TW201306552A (en) | 2013-02-01 |
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