M395968 五、新型說明· 【新型所屬之技術領域】 本創作係有關於一種跨語音平台之通訊系統，更詳而 言之，係一種透過對話啟動協定（Session Initiation Protocol，SIP)伺服器以供採用公眾交換電話網路或行動 通訊網路進行通訊的發話端與受話端進行連線通訊之跨語 音平台之通訊系統。 【先前技術】 隨著網際網路蓬勃發展，通訊不再侷限於傳統電話的 撥接方式，像是網路電話，僅需簡易設備及網路，即可讓 使用者在達到通訊節費的同時兼具維持一定的通話品質， 因而網路電話在通訊市場佔有一席之地，遂成為企業公司 作為通訊聯絡的主要選擇。 雖網路電話具有節費的優點，但仍有部分問題無法克 服，舉例來說，公司内部設置網路電話，但公司員工外出 而處於無法與網際網路連結的通訊環境下，該公司員工僅 能透過傳統電話作聯繫，若該公司員工與欲通話的對方處 於同一區域或縣市，則所產生的通訊費用不會太多，但若 通話雙方分別處於不同縣市，甚至分別處於不同國家，則 可能產生驚人的通訊費用，此時公司内部所設置的網路電 話也無法達到節費效果。 此外，對於異質語音平台間的通訊處理，以目前常見 的Skype即時通訊軟體而言，安裝有該Skype即時通訊軟 體的網路終端，例如電腦，雖該電腦藉由Skype即時通訊 M395968 軟體即可提供如網路電話般的撥打功能而撥打到市話、行 動電話或國際電話，但仍僅限於以網路電話般的撥打功能 撥打到傳統電話機上，而無法供傳統電話機或行動電話撥 打到安裝有該Skype即時通訊軟體的網路終端。因此，只 要發話端處於無法與網際網路連結的通訊環境從而無法使 用網路電話資源的情況下，發話端就無法達到節費效果。 【新型内容】 鑒於上述習知技術之缺點，本創作目的在於提供一種 跨語音平台之通話系統，用以供採用公眾交換電話網路或 行動通訊網路進行通訊的發話端與受話端建立連線並通 訊，藉此使非處於網路電話語音平台下進行通訊服務的發 話端亦可達到節費的效果。 為達前述目的及其他目的，本創作提供一種跨語音平 台之通訊系統，係透過對話啟動協定伺服器以供採用公眾 交換電話網路或行動通訊網路進行通訊的發話端與受話端 建立連線並通訊，包括：資料庫單元，係儲存語音資料及 使用者資訊；語音單元，係用於接收該發話端所輸入之輸 入資訊，且依據該輸入資訊提供對應之語音資料至該發話 端；處理單元，係用於解析該輸入資訊以由該資料庫單元 取得對應之使用者資訊，並產生連線訊息，以供該對話啟 動協定伺服器依據該連線訊息向受話端提出連線請求；以 及通訊單元，係用以將回應允准連線的受話端與該發話端 建立連線以進行雙方通訊。 於一實施形態中，該發話端之語音平台係採用公眾交 5 M395968 之語音平台係採 換電話網路或行―_路；㈣受_ 用網際網路或虛擬私有網路之網路通訊。 此外 必处埋早元復包括驗證模組，以 俾確認該發話端之使用者資訊。 &供驗*機制 相較於習知技術，本創作之跨語音平 為發話端與網路電話語音平台的通訊橋樑:藉㈣跨= 平台=通=统使採用絲交換電話網路或行動通訊網ς 進灯通5fun端，透過對話啟動協定伺服器即可與受話 端以網路電話通訊方式建立連線並通訊，藉以節省發話端 電信費用。 【實施方式】 —以下係藉由特定的實施形態說明本創作之技術内 容，熟悉此技藝之人士可由本說明書所揭示之内容輕易地 瞭解本創作之其他優點與功效。本創作亦可藉由其=不同 的實施形態加以施行或應用。 請參閱第1圖，係本創作跨語音平台之通訊系統一實 把形悲下的糸統基本架構以及其應用架構示意圖。如圖所 示，該跨語音平台之通訊系統1係透過對話啟動協定 (Session Initiation Protoco卜 SIP)伺服器 40 提供採用公 眾交換電話網路（public switched telephone network， PSTN)或行動通訊網路（Mobile Network)進行通訊的發 話端100可與受話端200以網路電話通訊方式建立連線並 通訊。 該SIP伺服器目前被用於多方多媒體通訊（Multiparty M395968M395968 V. New Description · [New Technology Area] This creation is about a communication system across voice platforms. More specifically, it is a Session Initiation Protocol (SIP) server for adoption. A cross-voice platform communication system in which the public exchange telephone network or mobile communication network communicates with the caller and the caller. [Prior Art] With the rapid development of the Internet, communication is no longer limited to the dial-up mode of traditional telephones. For example, Internet telephony requires only simple devices and networks to allow users to reach communication fees. Both of them maintain a certain quality of call, so Internet telephony has a place in the communications market, and it has become the main choice for corporate companies as a communication link. Although Internet telephony has the advantage of saving money, there are still some problems that cannot be overcome. For example, the company has set up an internal network phone, but the company employees go out and are in a communication environment that cannot be connected to the Internet. Can communicate through traditional telephones. If the employees of the company are in the same area or county or city, the communication costs will not be too much, but if the two parties are in different counties and cities, or even in different countries, There may be an alarming communication fee, and the Internet phone set up inside the company cannot achieve the effect of the fee. In addition, for the communication processing between heterogeneous voice platforms, the current Skype instant messaging software, the network terminal installed with the Skype instant messaging software, such as a computer, can be provided by the Skype instant messaging M395968 software. Dialing to a local, mobile or international call, such as a VoIP call, but still limited to VoIP-like dialing to a traditional telephone, but not for a traditional telephone or mobile phone. The Skype instant messaging software network terminal. Therefore, in the case where the originating terminal is in a communication environment incapable of connecting to the Internet and thus cannot use the VoIP resources, the calling terminal cannot achieve the saving effect. [New content] In view of the shortcomings of the above-mentioned prior art, the purpose of the present invention is to provide a cross-voice platform communication system for connecting a call end and a call end of a communication using a public switched telephone network or a mobile communication network. Communication, so that the end of the communication service that is not under the VoIP voice platform can also achieve the effect of the fee. For the above purposes and other purposes, the present invention provides a communication system for a cross-speech platform, which initiates an agreement server through a dialogue for establishing a connection between a caller and a callee using a public switched telephone network or a mobile communication network for communication. The communication includes: a database unit for storing voice data and user information; and a voice unit for receiving input information input by the caller, and providing corresponding voice data to the caller according to the input information; For parsing the input information to obtain corresponding user information by the database unit, and generating a connection message for the session initiation agreement server to make a connection request to the receiver according to the connection message; and communication The unit is used to establish a connection between the called end of the response and the calling end for communication between the two parties. In one embodiment, the voice platform of the utterance end uses a voice platform of the public to exchange 5 M395968 to exchange telephone network or line _ road; (4) to communicate by _ using the Internet or virtual private network. In addition, it is necessary to bury the early recovery module including the verification module to confirm the user information of the utterance. Compared with the prior art, the cross-voice of this creation is the communication bridge between the utterance and the VoIP voice platform: borrowing (four) span = platform = pass = unified use of silk exchange telephone network or action The communication network ς enters the 5fun end of the light, and initiates the agreement server through the dialogue to establish a connection and communication with the receiving end by means of network telephone communication, thereby saving the telecommunications cost of the calling terminal. [Embodiment] The following is a description of the technical content of the present invention by a specific embodiment, and those skilled in the art can easily understand other advantages and effects of the present invention by the contents disclosed in the present specification. This creation can also be implemented or applied by its different implementations. Please refer to Figure 1, which is a schematic diagram of the basic architecture of the communication system of the cross-speech platform and its application architecture. As shown in the figure, the communication system 1 of the cross-voice platform provides a public switched telephone network (PSTN) or a mobile communication network (Mobile Network) through a Session Initiation Protocol (SIP) server 40. The transmitting terminal 100 for communication can establish a connection and communication with the receiving terminal 200 by means of network telephone communication. The SIP server is currently used for multi-party multimedia communication (Multiparty M395968)
Multimedia Communications)上，以作為網路端點間溝通 載具，因此，本創作之跨語音平台之通訊系統丨欲將發話 端100與受話端200建立連線時，仍需透過SIp伺服器的 進打連線及通話請求，例如身分認證、由所指定位址找出 受話端的位址或是通話管理等，由於該SIp飼服器4〇為當 前網路電話常用的設備，故以下不為文贄述’僅說明與本 創作相關之部分。本_跨語音平台之軌純包括：、 料庫單元―η、語音單s 12、處理單元13以及通訊單元… 該資料庫單元η係儲存語音資料及使用者資訊 言之，該資料庫單元η用於儲存該跨語音平台之通訊系殊 !的育料’該語音資料為該跨語音平台之通㈣統】 預設語音對話，舉财說，語音内容可能為“請制者輪1 代號”或“選擇功能選項，，等，藉以提供該跨語音平台之通訊 糸統1與使用者（特別是發話端1⑻）間互動；該使用者 貪訊係指發話端刚或受話端200之使用者資訊，主要用 於確3忍使帛者㈣’以供後續如紐或帳務時❹ 後詳述。 4 該語音單元12係用於接收該發話端⑽之輸 A ’且依^輸人資訊提供對應之語音資料至該發話端 Τ此階段主要是當發話端！⑼撥話至該跨語音平台之通 系’充N· &方進行需求輸入及/或資料驗證，舉例來 說^跨語音平台之通訊系統1會提供語音資料以提示發 話端loot使用者輸入所需資料而發話端依據該二 音貢料而回應並輸人欲通訊之受話端的公司代號、: 7 M395968 機或疋人員代碼等資料’總言之，該語音單元12是提供該 跨語音平台之通訊系統 1與該發話端100間的訊息傳遞。 再者’該輪入資訊係包括語音訊息或數位訊息。意即 由發話端1〇〇所輸入的輸入資訊可為語音或數位方式來進 仃，该語音訊息即是發話端100之使用者透過語音進行資 料輸入，此時該跨語音平台之通訊系統丨對應提供語音分 析，°亥數位訊息則是透過電話按鍵所產生，例如依據語音 提不刼作傳統電話機的按鍵以輸入欲撥打的電話號碼，而 產生對應該電話號碼的數字代碼。 °亥處理單元13係用於解析該輸入資訊以由該資料庫 ^ " 1取得對應之使用者資訊，並產生連線訊息。該處理 二=3主要是將跨語音平台之通訊线丨所接收之輸入資 么進彳解析，以知悉該發話端1⑽所欲連線對象，即欲連 線的受話端20G’由於該輸人#訊可能僅為前述的數字代 因而需由該資料庫單元丨丨取得對應的使用者資訊，並 自生用於將發話端_與受話端綱作通訊連線之連線訊 :SIP㈤服器40係依據該處理單元13所產生的連線 向魏端200提出連線請求，以使該通訊單元14可依 回應允准連線的受話端得與該發話端建立連線以進 方通訊。 又 路雷^體而言’該發話端⑽所使用的通訊設備並非是網 ^，而是如行動電話或市話等之傳統電話機，缺 由本創作跨語音平台之㈣純】可成為發話端⑽與^ M395968 • 路電話語音平台間的連結橋樑，藉此使發話端隨時隨地使 用網路電話資源以達到節費的效果，特別是在發話端100 處於無法與網際網路連結的環境從而無法使用網路電話資 源的情況下。 於一具體實施例中，而該受話端200之語音平台係採 用網際網路（Internet )或虛擬私有網路（virtual private network，VPN)之網路通訊架構。於此實施例中，該發話 端100之使用者是處於未設置有前述網路通訊架構的環境 ® 下，因此，無法透過費用便宜的網路電話與受話端200聯 繫，而僅能透過公眾交換電話網路或行動通訊網路等傳統 電話機制撥話至受話端200，此時會產生較高通話費用而 無法達到節費之情況。故，透過本創作之跨語音平台之通 訊系統1可使如行動電話或市話等傳統電話與受話端以網 路電話方式進行通話。 此外，該資料庫單元11儲存有一對應表，用於記錄 0 連線代碼及與該連線代碼對應的使用者資訊，以供該處理 -單元13透過該對應表以取得該受話端之使用者資訊，而該 連線代碼即用以供處理單元13與該發話端100所輸入之輸 入資訊進行比對，以在比對該輸入資訊與連線代碼一致 後，再進下一步取出與該連線代碼對應的使用者資訊。簡 單來說，針對該發話端100所輸入之輸入資訊，可由資料 庫單元11之對應表來找出相對應關係，不僅可取得發話端 100之使用者資訊，亦可找出受話端200之使用者資訊； 而該處理單元13即透過該對應表來取得該受話端200之使 9 M395968 用者資訊，並產生連線訊息，以供該SIP伺服器40依據該 處理單元13所產生的連線訊息向受話端200提出連線請 求，並由該通訊單元14將回應允准連線的受話端200與該 發話端100建立連線以進行雙方通訊。 再者，該語音單元12另可提供受話端200忙線時的 語音提示。具體來說，一般電話通訊或網路電話的撥打過 程中，當撥打對象忙線時，可能僅回應發話端1〇〇簡易的 聲響或者直接斷線的情況，反觀，透過該語音單元12可對 受話端200忙線時給予語音提示，例如“線路忙線中’’之類 的語音提示以通知發話端100。 如第2圖所示，係用以說明運用本創作跨語音平台之 通訊系統之發話端與受話端的具體實施例示意圖。 發話端透過傳統通訊設備50撥話至本創作跨語音平 台之通訊系統2，以藉由該跨語音平台之通訊系統2建立 發話端與受話端間的連線，該傳統通訊設備50依據本端通 訊服務透過公眾交換電話網路或行動通訊網路51作為其 通訊傳遞管道，該傳統通訊設備50經由語音單元22之語 音引導以將欲連線對象的受話端資料輸入，再透過處理單 元23分析判斷以產生連線訊息，以供SIP伺服器40將發 話端以及受話端雙方進行連線。 此外，受話端若處於有網路架構的通訊環境，因而可 使用網路電話設備60、70進行通訊，其中，該網路電話設 備60係可透過虛擬私有網路61作為通訊傳遞管道，例如 透過公司内部網路（Intranet)進行聯繫，而該網路電話設 備70係可透過網際網路7]作為其通訊傳遞管道 遠在國外的公㈣繫。因此，藉由本創作跨語音平台之通 :系統可使發話端的撥號處理不受限於受話端之網路電話 设備60、70所使用的網路架構。Multimedia Communications) is used as a communication carrier between network endpoints. Therefore, the communication system of the cross-voice platform of this creation wants to connect the sender 100 to the receiver 200, and still needs to pass through the SIp server. For connection and call requests, such as identity authentication, finding the address of the callee by the specified address, or call management, since the SIp feeder is the device commonly used for current Internet telephony, the following is not The narrative 'only describes the parts related to this creation. The track of the _ cross-speech platform includes:, the library unit η, the voice list s 12, the processing unit 13 and the communication unit. The database unit η stores voice data and user information, and the database unit η The communication material used to store the communication system of the cross-speech platform is 'the voice data is the communication of the cross-speech platform (four) system】 Preset voice dialogue, the money said, the voice content may be "requester wheel 1 code" Or "select a function option," to provide interaction between the communication system 1 and the user (especially the caller 1 (8)); the user's greed refers to the user of the utterer or the receiver 200 The information is mainly used to confirm the 3 (4) 'for subsequent follow-up or account accounting. 4 The voice unit 12 is used to receive the A' of the caller (10) and according to the information. Provide corresponding voice data to the caller. At this stage, it is mainly when the caller is sent! (9) Dial the call to the cross-voice platform's 'charged N· & party for demand input and/or data verification, for example ^ cross Voice platform communication system 1 will provide voice The company code is used to prompt the user to input the required information, and the sender responds according to the two-tone tribute and enters the company code of the receiver of the communication, 7 M395968 machine or employee code, etc. The voice unit 12 is a message transmission between the communication system 1 and the utterance terminal 100. The 'wheeling information system includes a voice message or a digital message. That is, the input input by the utterer 1 〇〇 The information can be entered in a voice or digital manner. The voice message is the data input by the user of the voice terminal 100. At this time, the communication system of the cross-voice platform provides voice analysis, and the digital information is transmitted through the The telephone button is generated, for example, according to the voice of the traditional telephone to input the telephone number to be dialed, and the digital code corresponding to the telephone number is generated. The processing unit 13 is configured to parse the input information for the data. The library ^ " 1 obtains the corresponding user information and generates a connection message. The processing 2 = 3 is mainly to connect the communication line across the voice platform The input of the input is analyzed, so as to know the object to be connected to the terminal 1 (10), that is, the terminal 20G of the line to be connected, because the input may only be the aforementioned digital generation, and thus the database unit is required.丨 Obtain the corresponding user information, and self-generated to connect the utterer _ with the receiver interface communication: SIP (five) server 40 is based on the connection generated by the processing unit 13 to the Wei end 200 The line request is such that the communication unit 14 can establish a connection with the calling terminal to communicate with the incoming terminal according to the answering line of the permitted connection. In addition, the communication device used by the transmitting terminal (10) is not Network ^, but a traditional telephone such as mobile phone or local telephone, the lack of the cross-voice platform of this creation (four) pure] can become a bridge between the caller (10) and ^ M395968 • road phone voice platform, so that the caller is always available Use VoIP resources anywhere to achieve savings, especially if the endpoint 100 is in an environment that is not connected to the Internet and cannot use VoIP resources. In a specific embodiment, the voice platform of the receiver 200 adopts a network communication architecture of the Internet or a virtual private network (VPN). In this embodiment, the user of the utterance terminal 100 is in an environment® that is not provided with the foregoing network communication architecture, and therefore cannot communicate with the receiver 200 through a cheap Internet phone, but only through the public exchange. A traditional telephone mechanism such as a telephone network or a mobile communication network dials the call to the caller 200, which results in a higher call charge and cannot reach the fee. Therefore, the communication system 1 of the cross-voice platform of the present invention enables a conventional telephone such as a mobile phone or a local telephone to make a telephone call by means of a network telephone. In addition, the database unit 11 stores a correspondence table for recording the 0 connection code and the user information corresponding to the connection code, so that the processing unit 13 transmits the correspondence table to obtain the user of the called terminal. Information, and the connection code is used for comparing the input information input by the processing unit 13 with the utterance terminal 100, and then, after the input information is consistent with the connection code, the next step is taken out. User information corresponding to the line code. Briefly, the input information input by the utterance terminal 100 can be found by the correspondence table of the database unit 11 to obtain the corresponding relationship, not only the user information of the utterance terminal 100 but also the use of the receiver terminal 200. The processing unit 13 obtains 9 M395968 user information of the receiving terminal 200 through the correspondence table, and generates a connection message for the SIP server 40 to generate a connection according to the processing unit 13. The message is sent to the receiving end 200 to make a connection request, and the communication unit 14 establishes a connection between the receiving terminal 200 that responds to the permitted connection and the calling terminal 100 to perform communication between the two parties. Moreover, the voice unit 12 can further provide a voice prompt when the called terminal 200 is busy. Specifically, during the dialing process of a general telephone communication or a network telephone, when the busy line of the object is dialed, it may only respond to the simple sound of the calling terminal or the direct disconnection. In contrast, the voice unit 12 can be used to When the receiving terminal 200 is busy, a voice prompt is given, for example, a voice prompt such as “in line busy line” to notify the calling terminal 100. As shown in FIG. 2, it is used to describe the communication system using the creative cross-speech platform. A schematic diagram of a specific embodiment of the calling terminal and the receiving end. The calling terminal dials the traditional communication device 50 to the communication system 2 of the cross-speech platform to establish a connection between the calling terminal and the receiving terminal through the communication system 2 of the cross-voice platform. In the line, the conventional communication device 50 serves as a communication transmission pipeline through the public switched telephone network or the mobile communication network 51 according to the local communication service, and the conventional communication device 50 is guided by the voice of the voice unit 22 to connect the terminal of the object to be connected. The data is input and analyzed by the processing unit 23 to generate a connection message for the SIP server 40 to perform both the calling terminal and the receiving terminal. In addition, if the receiving end is in a communication environment with a network architecture, the VoIP device 60, 70 can be used for communication, wherein the VoIP device 60 can communicate as a communication transmission channel through the virtual private network 61. For example, through the company's internal network (Intranet), the Internet telephony device 70 can be used as the communication transmission pipeline of the Internet (4) through the Internet 7]. Therefore, through this creative cross-voice platform Communication: The system can make the dialing process of the calling terminal not limited to the network architecture used by the network telephone devices 60, 70 of the receiving end.
:第：圖：示，配合前述第】圖所示的應用架構說明 |H乍之平台之通㈣統與發話端、受話端以及 伺服益間的電話撥打處理時序圖。如圖所示，首先執 ^時序W係指發話端⑽撥打電話至本創作跨語音平 =通訊系統！，並且輸人服務代碼，該服務代碼意指發 。舌端100欲進行網路電話撥打服務。 接著進•序τ2，其係指該跨語音平台之通訊系統1 =應語音衫至發話端⑽並特其喊，該語音提示可 成為：請使用者輸人欲連線之虛擬網路代碼。 接著進行時序Τ3，其仙發話端⑽輸人欲連線之虛 擬網路代碼。 鲁接著進行時序了4’其係指該跨語音平台之通訊系統】 應語音提示至發話端⑽並科其喊，該語音提示可 月b為.凊使用者輸入欲連線之公司簡碼。 接著進仃時序丁5’其係指發話端100輸入欲連線之公 司簡碼。 接著進订時彳T6 ’其為該跨語音平台之通訊系統1 °舌縞100之需求’轉而向SIP伺服器40提出連線及 通話請求。 接著進行4序丁7 ’其則是該Slp伺服器4〇向受話端 M395968 200提出連線及通話請求。 接著進行時序T8，其則是說明發話端100與受話端 200產生連線。 再者，若該發話端100與該受話端200分屬不同語音 平台時，亦即該發話端100屬於公眾交換電話網路或行動 通訊網路，而該受話端200屬於網際網路或虛擬私有網 路，以致於一端發出的訊號無法直接傳遞至另一端，故， 兩端中間仍需透過跨語音平台之通訊系統2作為中介橋 樑。此外，發話端100僅需撥打傳統電話至跨語音平台之 通訊系統1中，而後續連線至受話端的處理則為網路電話 之訊號傳輸，如此可達到節費的效果，特別是受話端200 與發話端1〇〇距離遙遠時，如不同國家，發話端所節省下 的通話費用更是可觀。 請參閱第4圖，係本創作跨語音平台之通訊系統另一 實施形態下的系統基本架構以及其應用架構示意圖。本實 施形態與第1圖不同處在於：跨語音平台之通訊系統3復 包括具有訊號轉換模組341的通訊單元34以及具有驗證模 組331的處理單元33，以下僅就前述不同處提出說明。 該處理單元33之驗證模組331係提供驗證機制以確 認發話端100之使用者資訊。詳言之，該驗證模組331用 以對發話端1〇〇進行身分驗證，如此可對於通訊結果產生 帳務紀錄，舉例來說，在確認發話端100身分後，可方便 對此次通訊產生紀錄以供後續帳務處理。此外，透過該驗 證模組331可使受話端200設定安全機制，即該驗證模組 M395968 -331可由受話端200預設發話端100之使用者資料，藉此 避免不被允許之發話端的來電。因此，可透資料庫單元31 所儲存的使用者資料及相關驗證資訊來進行驗證，其中， 該驗證機制可為密碼辨識、語音辨識、該發話端之裝置認 證的任一者或其組合。 該通訊單元34的訊號轉換模組341用於將該發話端 • 100與該受話端200所傳遞之語音訊號進行類比訊號及數 -位訊號間的轉換。由於該發話端100與受話端200分屬不 •同語音平台，因此，兩者間所傳遞語音訊號格式可能不相 同，像是傳統通訊格式為類比訊號，而數位電話之訊號則 為數位格式，因此，透過該訊號轉換模組341可將該語音 訊號格式進行類比和數位間的轉換。 相較於習知技術，本創作跨語音平台之通訊系統為發 話端與網路電話語音平台間的連結橋樑，以提供採用公眾 交換電話網路或行動通訊網路進行通訊的發話端與受話端 | 間可以網路電話方式進行連線以及通話，藉此使發話端不 .因採用公眾交換電話網路或行動通訊網路撥打長途電話或 國際電話而產生高額的通訊費用；此外，本創作跨語音平 台之通訊系統除了透過語音引導讓發話端輸入所欲通訊連 線的受話端資料外，更提供驗證機制來提高連線安全性。 上述實施例僅例示性說明本創作之原理及其功效，而 非用於限制本創作。任何熟習此項技藝之人士均可在不違 背本創作之精神及範疇下，對上述實施例進行修飾與改 變。因此，本創作之權利保護範圍，應如後述之申請專利 13 範圍所列。 【圖式簡單說明】 第1圖係為本創作跨語音平台之通訊系統一實施形態 下的系統基本架構以及其應用架構示意圖； ⑴第2圖係為運用本創作跨語音平台之通訊系統之發話 令而與受话端的具體實施例示意圖； 第3圖係為配合前述第！圖所示的應用架構說明 作之跨語音平台之通訊系統與發話端、受話端以及幻 服器間的電話撥打處理時序圖；以及 ，第4圖係為本創作跨語音平台之通訊系統之另 形態下的系統基本架構以及其應用架構示意圖。 、也 【主要元件符號說明】 1、2、3 跨語音平台之通訊系統 11、31 資料庫單元 12、22 語音單元 13、23、33 處理單元 14、34 通訊單元 33] 驗證模組 341 訊5虎轉換模組 40 SIP伺服器 50 傳統通訊設備 51 公眾交換電話網路或行動通 60、70 網路電話設備 61 虛擬私有網路 訊網路 M395968 71 網際網路 100 發話端 200 受話端 Τι~Τ8 時序: Pd: Figure: Shows the application architecture description shown in the above figure] |H乍's platform communication (4) system and caller, receiver and servo benefit telephone dialing processing timing diagram. As shown in the figure, first, the timing W is the calling terminal (10) to make a call to the creative cross-voice flat = communication system! And enter the service code, which means send. The tongue end 100 is intended to make a telephone call service. Then, the order τ2, which refers to the communication system of the cross-voice platform 1 = should be voiced to the speech terminal (10) and specifically called, the voice prompt can become: the user enters the virtual network code to connect. Then, the timing Τ3 is performed, and the sacred terminal (10) inputs the virtual network code to be connected. Lu then proceeded with the sequence 4', which refers to the communication system of the cross-voice platform.] The voice prompt should be sent to the caller (10) and the voice call can be made. The voice prompt can be the monthly b. The user enters the company shortcode to be connected. Then enter the time sequence D'5, which means that the speech terminal 100 inputs the company short code to be connected. Then, at the time of ordering, 彳T6', which is the demand of the communication system 1° tongue and groove 100 of the cross-speech platform, is forwarded to the SIP server 40 to make a connection and a call request. Then, the sequence 7 and the call request are made to the receiver M395968 200. Next, a sequence T8 is performed, which illustrates that the calling terminal 100 and the terminating end 200 are connected. Furthermore, if the calling terminal 100 and the receiving terminal 200 belong to different voice platforms, that is, the calling terminal 100 belongs to a public switched telephone network or a mobile communication network, and the called terminal 200 belongs to the Internet or a virtual private network. Therefore, the signal sent from one end cannot be directly transmitted to the other end. Therefore, the communication system 2 across the voice platform still needs to be used as an intermediate bridge between the two ends. In addition, the calling terminal 100 only needs to dial the traditional telephone to the communication system 1 of the cross-voice platform, and the subsequent connection to the receiving end is the signal transmission of the network telephone, so that the effect of the fee can be achieved, especially the receiving terminal 200. When the distance from the caller is far away, such as in different countries, the cost of the call saved by the caller is considerable. Please refer to FIG. 4, which is a schematic diagram of the basic architecture of the system and its application architecture in another embodiment of the communication system of the cross-voice platform. The present embodiment differs from the first embodiment in that the communication system 3 across the voice platform includes a communication unit 34 having a signal conversion module 341 and a processing unit 33 having a verification module 331. Hereinafter, only the differences will be described. The verification module 331 of the processing unit 33 provides an authentication mechanism to confirm the user information of the utterance terminal 100. In detail, the verification module 331 is configured to perform identity verification on the utterance end, so that a accounting record can be generated for the communication result. For example, after confirming the identity of the utterance terminal 100, the communication can be conveniently generated. The record is for subsequent accounting. In addition, the authentication module 331 can be used to set the security mechanism for the receiving terminal 200. That is, the authentication module M395968-331 can preset the user data of the calling terminal 100 by the receiving terminal 200, thereby avoiding the incoming call of the calling terminal. Therefore, the verification can be performed through the user data stored in the database unit 31 and the related verification information, wherein the verification mechanism can be any one of a combination of password identification, voice recognition, device authentication of the calling terminal, or a combination thereof. The signal conversion module 341 of the communication unit 34 is configured to convert the speech signal transmitted between the speech terminal 100 and the receiving terminal 200 by analog signals and digital-bit signals. Since the speech terminal 100 and the receiving terminal 200 belong to the same voice platform, the voice signal format transmitted between the two may be different, such as the traditional communication format is analog signal, and the digital telephone signal is in digital format. Therefore, the voice signal format can be converted into analog and digital bits by the signal conversion module 341. Compared with the prior art, the communication system of the cross-speech platform is a bridge between the speech terminal and the voice telephone platform of the Internet, and provides a communication terminal and a receiver terminal for communication using a public switched telephone network or a mobile communication network. Internet connection can be used to make connections and calls, so that the caller does not have a high communication fee due to the use of the public switched telephone network or the mobile communication network to make long distance calls or international calls. In addition, the creative cross-voice platform In addition to voice input, the communication system allows the input terminal to input the data of the connected terminal of the desired communication connection, and provides an authentication mechanism to improve the connection security. The above embodiments are merely illustrative of the principles of the present invention and their effects, and are not intended to limit the present invention. Any person skilled in the art can modify and change the above embodiments without departing from the spirit and scope of the present invention. Therefore, the scope of protection of this creation should be as listed in the scope of Patent Application 13 described later. [Simple description of the diagram] The first diagram is a schematic diagram of the basic architecture of the system and the application architecture of the communication system based on the creation of the cross-voice platform; (1) The second diagram is the speech system using the communication system of the cross-voice platform. A schematic diagram of a specific embodiment of the command and the receiving end; the third figure is to cooperate with the foregoing! The application architecture shown in the figure illustrates the timing diagram of the telephone dialing process between the communication system of the cross-voice platform and the calling terminal, the receiving end, and the phantom server; and, FIG. 4 is another communication system for creating a cross-voice platform. Schematic diagram of the basic architecture of the system and its application architecture. [also, main component symbol description] 1, 2, 3 cross-voice platform communication system 11, 31 database unit 12, 22 voice unit 13, 23, 33 processing unit 14, 34 communication unit 33] verification module 341 5 Tiger conversion module 40 SIP server 50 Traditional communication device 51 Public switched telephone network or mobile communication 60, 70 Internet telephone equipment 61 Virtual private network communication network M395968 71 Internet 100 speech terminal 200 Receiver terminal Τι~Τ8 Timing