200847748 九、發明說明： 【發明所屬之技術領域】 本發明係關於一種網路：！話系統，特別是關於一種可 同時登錄多個帳號之會議初始通信協定模式(SIp)網路電 話系統。〜 【先前技術】 在現今的語音通訊領域裡，網路電話(v〇ice 〇ver Internet Protocol，VoIP)幾乎已是不可或缺的主要組成。網 路電話的基本原理利用語音壓縮及編/解碼的方式將類比 的語音編碼並壓縮成數位語音訊號，再藉由網際網路協定 (Internet Protocol)將此數位化語音訊號以資料的形式上傳 至網際網路。 常見的VoIP的通訊協定種類有h.323(ITU-T KecGmmeiidation EL323)、媒體閘道控制協定 Gateway Control Protocol ’ MGCP)、會議初始通信協定 (Session Initiation Protocol，SIP)以及 SKYPE 通訊協定等。 其中，以SIP以及SKYPE這兩種通訊協定，近年來最為 普遍以及廣受歡迎。 SIP已經被公認為網際網路與傳統電信公共電話交換 網路(PSTN)整合的關鍵技術，而且它也被第三代行動通訊 系統(3G)採用為未來無線多媒體通訊的技術標準。sip支 援網際網路協定(IP)、企業用網路交換200847748 IX. Description of the invention: [Technical field to which the invention pertains] The present invention relates to a network:! A system, in particular, a conference initial protocol mode (SIp) network telephone system that can simultaneously log into multiple accounts. ~ [Prior Art] In today's voice communication field, VoIP (VoIP) is almost an indispensable component. The basic principle of VoIP uses voice compression and encoding/decoding to encode and compress the analog voice into digital voice signals, and then upload the digital voice signals as data through the Internet Protocol (Internet Protocol). Internet. Common VoIP communication protocols include h.323 (ITU-T KecGmmeiidation EL323), Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), and SKYPE protocol. Among them, the two communication protocols, SIP and SKYPE, have become the most popular and popular in recent years. SIP has been recognized as a key technology for the integration of the Internet with the traditional telecommunications public switched telephone network (PSTN), and it has also been adopted by the third generation mobile communication system (3G) as the technical standard for future wireless multimedia communications. Sip support Internet Protocol (IP), enterprise network exchange
Pnvate Branch exchange)、3G及網路電信交換機 (Soft-switch)，並可用於多媒體服務，極有成為ν〇ΙΡ技術 5 200847748 中最重要的協定之一 由於SIP簡潔的技術與其所具有可 、 配合環境的彈性，使該協定極適合在網際網路環境中使 . 用，這使得SIP協定的使用在企業體中有逐漸增加的趨勢。 一般SIP無線網路電話的基本架構包含了兩個部分·· 一個主機(Base)和複數初手持式子機(Handset)。其中複數 個手持式子機透過數位增強無線電詰通信協議(Digital Enhanced Cordless Te]ecommunicati〇ns，DECT)註冊到主 機，再藉由主機通過SIP協議實現網路電話的功能。然而， Ρ 目前這種架構的SIP無線網路電話一次只能註冊一個SIp 帳號，因此，複數個手持式子機在同一時間内只能共同使 用一個SIP帳號的>^息，即在同一時間内只能有一個手持 式子機在通話；這使得一個主機雖然能和複數個手持式子 機透過無線協定連結，卻無法使一個主機同暗支持複數個 sn>帳號，即每-個手持式子機利用一個Slp帳號，同時 藉由主機通過SIP協議執行網路電話的功能實際發揮。 、目此S要-種可赠登鮮個親之網路電話系統及 方法’以實現在SIP無線網路電話系統中，多個手持式子 機可登錄不同帳號，並分別進行通話的功能。 【發明内容】 本發明之㈣在雖供—翻路電話祕，藉由在網 路電話主機及網路電話子機中增加管理模組，以達成網路 -電話主機同時登錄複數個帳號之網路電話。 1 本發鴨«露-種可同輕騎數娜狀網路電話 祕’至少包括：―伺服器網路電話主機，具有-用 6 200847748 戶管理模組、-子機管賴組與—帳號管理模組，藉由網 際網路連結伺服器；以及複數個網路電話子機，其中每一 ，網路電話子機皆具有-無線通鶴議模組、-帳號註冊 官理模組與’息管賴組，又任—個網路電話子機可藉 由，線通信協議模組_網路電話主機，經由帳號註冊^ 理核組註職聊可制。其巾，藉由帳號管理模組可同 時登錄並管理複__舰，麟每—伽叫電話子機 可註冊一個網路電話帳號，透過網路電話主機與伺服器遠 成連結。 、。% 、種網路電話糸統之操作方法，係適用於一可同時登 錄複數個帳號之網路電話系統，其中網路電話系統包含二 伺服器、一網路電話主機與複數個網路電話子機，該操作 方法包含以下步驟··姻路電話主機接收一第一子機註冊訊 號；網路電話主機與第一子機建立一無線數據通道;第一 子機發送一請求訊號以登錄網路通信協定帳號至網路電話 主機；網路電話主機接收請求訊號，並完成網路通信協定 帳號登錄；第一子機登錄成功後，發送一同步要求至網路 電話主機；以及網路電話主機回應第一子機要求並完成同 步。 【實施方式】 請參考圖一，係為本發明一實施例網路電話系統之示 意圖。本發明所揭露之網路電話系統1包括一會議初始通 信協定(SessionlnitiationProtocol，SIP)伺服器 1〇(以下簡稱 SIP伺服器）、一網路電話主機12與複數個網路電話子機 200847748 14。其中，網路電話主機12具有一帳號管理模組12〇、一 子機管理模組12·2與一用戶管理模組124，藉由網際網路 可連結SIP伺服器1〇 ;任一個網路電話子機(以第一子機 Ml舉例說明）皆具有一無線通信協議模組141〇、一帳號往 冊管理挺組1412與一信息管理模組141伞，藉由有線^無 線通信協議模組1410連結網路電話主機，經由帳號註 冊管理模組1412註冊通過即可使用。其中，藉由帳號管理 模組120可同時登錄並管理複數個網路帳號，使得每一個 網路電話子機14可註冊一個網路電話帳號，透過網路電話 主機12與SIP伺服器10達成連結。 帳號管理模組120係負責管理網路電話主機12中目前 已經登錄之sip帳號信息，且其主要工作乃是維持SIp帳 號與SIP伺服器10之間的連結。當網路電話主機12接收 到由SIP伺服器10發送的一，SIP電話接聽(通)要求時， 帳號管理模組120接收接聽要求的連結，並維持此一接聽 要求與該帳號之間連結的通暢；當網路電話主機12接收到 弟一子機的撥號要求時’帳號管理模組120負責將第一 子機141所屬的SIP帳號的信息以及要撥打的帳號信息發 送到SIP伺服器10，請求SIP伺服器10建立連結，當連 結建立後，負責雙方的語音以及其他的數據封包能夠正確 傳輸。 子機管理模組122負責管理網路電話子機14的無線通 "ί吕協議(DECT)註冊與註銷；包含所有已經註冊成功網路電 話子機14的信息’網路電話子機14與網路電話主機12 之間無線通道的建立、維持及數據傳輪。其中，每一個網 200847748 路電話子機14皆錢過—個唯-的身份㈣標識。 用戶管理模組124負責管理網路電話系統i中的用戶 化息’其中每—細戶皆對應，個唯-的網路電話子機身 份標識和SIP帳號。用戶管理模組124主要是負責將一無 線通道與㈤路連線連接起來形&網路電話子機14到 SIP伺服器12的-條數據通路，如：第一子機叫與網路 電話主機12間的無線通道！，以及網路電話主機12至$正 舰器10之間的網路連線1;此一數據通路負責傳輸語言 和其它數據信息’以實現網路電話的接收和撥打，另外用 戶管理模組I24還負責管理對應SIP帳號的聯絡人信息， 帳號本身的設置信息，如此就可以實現不同的SIp用戶管 理自己獨有的聯絡人。 網路電話子機中的無線通信協議模組141〇，係用以負 貝無線通信協議註冊，維護無線通信協議通道的請求、建 立和數據的傳輸。 帳號§主冊管理模組1412 ’係用以接收幻p用戶輪入的 SIP帳號以及相關的設置信息，通過無線通道丨將信息傳 送到網路電話主機12。 信息管理模組1414，係用以管理SIP帳號的聯繫人信 息與相關設置信息等；這些信息將藉由網路電話子機14 連結到網路電話主機12時’經由網路電話主機同步傳 輪至網路電話子機14 ’而SIP用戶若在網路電話子機 端變更該些信息，則網路電話子機14會將變更同步傳輸至 網路電話主機12端。 9 200847748 在本發明網路電話系統1中，不同的網路電話子機14 …罔路電主機12間轉獨立的無線數據通道，不同的 S1P帳號與SIP伺服器10維持獨立的網路連線，如：第一 子機141與網路電話主機12之間的無線通道i，以及網路 電話主機12至SIP伺服器1〇之間的網路連線丨；第二子 機142與網路電話主機12之間的無線通道2，以及網路電 話主機12至SEP伺服器1〇之間的網路連線2 ;與第n子 機143與網路電話主機12之間的無線通道n，以/及網路電 話主機12至SIP伺服器10之間的網路連線n等等。 當第一 SIP用戶Ul·在第一子機141上登錄一個sip 帳號，需在網路電話主機12中通過SIp帳號與網路電話子 機身份標識的唯一識別，才能透過網路電話主機12連線到 sip饲服器ίο，以維持用戶m在網路電話系統i中獨立 的無線通道1與網路連線〗。當第二SIp用戶112在第二子 機142上登錄一個SIP帳號，需在網路電話主機12中通過 SIP帳號與網路電話子機身份標識的唯一識別，才能透過 網路電話主機12連線到sip伺服器10，以維持另一用戶 U2在網路電話系統1中獨立的無線通道2與網路連線2。 如此’不同的用戶可以在網路電話系統1中維持獨立 的數據通路’即不同的用戶在網路電話系統1維持獨立的 無線通道與網路電話主機12通訊，網路電話主機12藉由 獨立的網路連線與SIP伺服器10通訊，彼此相互獨立；故 本發明網路電話系統1可以藉由一個網路電話主機12同時 提供多個sip用戶使用網路電話子機14獨立通話。 200847748 請參考圖二’係為本發明一實施例網路電話系統連結 之流程圖。步驟S200，分別將網路電話主機與網路電話 ，各自開機。步驟S21G ’網路電話子機藉由無線通信協議 杈組魏-珊信息至:_電話域。轉S22Q，網路電 話主機的子機管理模組接枚註冊信息，並驗證網路電話子 機之身分標識以完成連結，建立一條無線通道。 步驟S212，一用戶於網路電話子機中輸入SIp帳號， 網路電話子機的帳號註冊管理模組接吹用續人之sip帳 號’發送-登錄要求至網路電話主機，請求登錄⑽帳號。 步驟S222 ’網路電話主機的帳號管理模組接收登錄要.^， 以完成SIP帳號到SIP伺服器的註冊，建立網路電話主 與SHM司服器之間的網路連線，並連結網路連線與無線通 道之數據通路。.步驟S224 ,當數據通道建立後，網路電話 主機發送一登錄成功之訊息至網路電話子機。 步驟S214，網路電話子機接收登錄成功之訊息，將由 仏息官理模組發送一同步請求之訊息至網路電話主機，請 奉網路電話主機端同步聯絡人及相關設置信息。步^ S226，網路電話主機的用戶管理模組接收同步請求之訊 ,息，並完成同步聯絡人及相關設置信息的動作。當同步動 作完成後，則網路電話系統即連結完成。 ^ 前述流程步驟簡單說明一網路電話系統連結之方式， 僅舉單一網路電話子機建立數據通路之方法，由於本發明 網路電話之網路電話主機可同時提供多個SIp帳號登入， 複數個網路電話子機建立數據通路之原理與上述相同，故 在此不多加贅述。 200847748 請參考圖三A，係為本發明一實施例網路電話系統撥 號清求通話之流程圖。當網路電話系統連結成功後，網路 電話系統即可開始撥話或是接聽來電。步驟S31〇 ,使用者 欲撥打電話，考人欲_了對方之號碼，網路電話子機雜 發送一通話請求至網路電話主機端。步驟S32〇，網路電話 主機^&接收通話請求’並傳遞該通話請求至幻p伺服器， 等待SIP伺服器之應答。步驟S33〇，Slp伺服器回應該通Pnvate Branch exchange), 3G and network-based soft-switch, and can be used for multimedia services. It is one of the most important agreements in 2008 〇ΙΡ technology 5 200847748. Because SIP's simple technology is compatible with it. The flexibility of the environment makes the agreement very suitable for use in the Internet environment, which makes the use of SIP agreements gradually increase in the enterprise. The basic architecture of a typical SIP wireless network phone consists of two parts: a base and a complex hand-held handset. A plurality of handheld handsets are registered to the host through a Digital Enhanced Cordless Tee (ecommunicati〇ns, DECT), and the host uses the SIP protocol to implement the function of the network telephone. However, Ρ SIP wireless network phones of this architecture can only register one SIp account at a time. Therefore, multiple handheld handsets can only use one SIP account at the same time, that is, at the same time. There can only be one handheld handset in the call; this allows a host to connect to a plurality of handheld handsets via a wireless protocol, but cannot enable a host to support multiple sn> accounts, ie, each handheld. The slave uses a Slp account and the actual function of the VoIP is performed by the host through the SIP protocol. In order to realize the SIP wireless network telephone system, multiple handheld handsets can log in to different accounts and perform separate calls. SUMMARY OF THE INVENTION In the present invention, (4) in addition to the network phone host and the network phone sub-machine, the management module is added to the network phone host to simultaneously log in to the plurality of accounts. Road call. 1 The hair duck «露-种可同轻骑数娜状网络电话秘' includes at least: "server network phone host, with -20082747748 user management module, - child machine management group and - account management The module, through the Internet connection server; and a plurality of network telephone handsets, each of which has a wireless telephone handset module, an account registration system module and an The management group can also be used as a network telephone sub-machine. The line communication protocol module _ VoIP host can be registered via the account registration. With the account management module, the account management module can simultaneously log in and manage the complex __ship. The ri-ga-calling phone sub-machine can register an VoIP account and connect to the server through the VoIP host. ,. %, the operation method of the network telephone system is applicable to a network telephone system capable of simultaneously logging in multiple accounts, wherein the network telephone system includes two servers, one network telephone host and a plurality of network telephones. The operation method includes the following steps: • the marriage telephone host receives a first slave registration signal; the network telephone host establishes a wireless data channel with the first slave; and the first slave sends a request signal to log in to the network. The communication agreement account to the network telephone host; the network telephone host receives the request signal, and completes the network communication protocol account login; after the first child machine successfully logs in, sends a synchronization request to the network telephone host; and the network telephone host responds The first slave requests and completes the synchronization. [Embodiment] Please refer to FIG. 1, which is a schematic diagram of a network telephone system according to an embodiment of the present invention. The network telephone system 1 disclosed in the present invention comprises a Session Initialization Protocol (SIP) server 1 (hereinafter referred to as a SIP server), a network telephone host 12 and a plurality of network telephone handsets 200847748 14. The VoIP host 12 has an account management module 12, a child management module 12·2 and a user management module 124, and can connect to the SIP server through the Internet; any network The telephone sub-machines (exemplified by the first sub-machine M1) each have a wireless communication protocol module 141, an account management management group 1412 and an information management module 141 umbrella, and a wired wireless communication protocol module. The 1410 is connected to the VoIP host and can be used after being registered and registered by the account registration management module 1412. The account management module 120 can simultaneously log in and manage a plurality of network accounts, so that each of the network phone handsets 14 can register a network phone account, and the network phone host 12 and the SIP server 10 are connected. . The account management module 120 is responsible for managing the sip account information currently registered in the VoIP host 12, and its main job is to maintain the connection between the SIp account and the SIP server 10. When the VoIP host 12 receives the SIP phone answering (pass) request sent by the SIP server 10, the account management module 120 receives the connection requesting the request and maintains the connection between the answering request and the account. The account management module 120 is responsible for transmitting the information of the SIP account to which the first child machine 141 belongs and the account information to be dialed to the SIP server 10 when the network telephone host 12 receives the dialing request of the child machine. The SIP server 10 is requested to establish a link. When the link is established, the voice and other data packets of both parties can be correctly transmitted. The sub-machine management module 122 is responsible for managing the wireless communication protocol (DECT) registration and deregistration of the network telephone handset 14; including all the information of the registered successful network telephone handset 14 'the network telephone handset 14 and The establishment, maintenance and data transmission of the wireless channel between the VoIP host 12. Among them, each of the network 200847748 telephone sub-machines 14 are all over the money--identity (four) identification. The user management module 124 is responsible for managing the user information in the VoIP system i, where each of the small users corresponds to a unique network phone sub-body identification and SIP account. The user management module 124 is mainly responsible for connecting a wireless channel to the (five) way connection to form a data path of the network telephone handset 14 to the SIP server 12, such as: the first child machine and the network phone Wireless channel between the host 12! And the network connection 1 between the VoIP host 12 and the stern 10; this data path is responsible for transmitting language and other data information to enable the reception and dialing of the VoIP, and the user management module I24 It is also responsible for managing the contact information of the corresponding SIP account and the setting information of the account itself, so that different SIp users can manage their own unique contacts. The wireless communication protocol module 141〇 in the network telephone handset is used for registration of the negative wireless communication protocol, and maintains the request, establishment and data transmission of the wireless communication protocol channel. The account § masterbook management module 1412' is configured to receive the SIP account and the related setting information that the phantom p user has entered, and transmit the information to the VoIP host 12 through the wireless channel. The information management module 1414 is configured to manage contact information and related setting information of the SIP account; the information will be synchronized to the VoIP host 12 by the VoIP phone 14 To the VoIP handset 14' and the SIP subscriber changes the information on the VoIP handset, the VoIP handset 14 transmits the change to the VoIP host 12 simultaneously. 9 200847748 In the network telephone system 1 of the present invention, different network telephone sub-machines 14 ... switch to a separate wireless data channel, and different S1P accounts and the SIP server 10 maintain independent network connections. For example, the wireless channel i between the first child machine 141 and the network telephone host 12, and the network connection between the network telephone host 12 and the SIP server 1〇; the second child machine 142 and the network a wireless channel 2 between the telephone host 12, and a network connection 2 between the network telephone host 12 and the SEP server 1; and a wireless channel n between the nth slave 143 and the network telephone host 12, And / and the network connection between the network telephone host 12 and the SIP server 10, and so on. When the first SIP user Ul· logs in a sip account on the first child machine 141, it needs to be uniquely identified in the VoIP host 12 through the SIp account and the VoIP identity, in order to be connected through the VoIP host 12 Wire to the sip feeding device ίο to maintain the user m in the VoIP system i independent wireless channel 1 and network connection〗. When the second SIp user 112 logs in a SIP account on the second child machine 142, it needs to be uniquely identified in the VoIP host 12 through the SIP account and the VoIP identity, in order to connect through the VoIP host 12. To the sip server 10, to maintain another user U2 in the VoIP system 1 independent wireless channel 2 and network connection 2. Thus, 'different users can maintain an independent data path in the VoIP system 1', that is, different users maintain an independent wireless channel in the VoIP system 1 to communicate with the VoIP host 12, and the VoIP host 12 is independent. The network connection communicates with the SIP server 10 and is independent of each other; therefore, the VoIP system 1 of the present invention can simultaneously provide a plurality of sip users by a VoIP host 12 to independently talk using the VoIP handset 14. 200847748 Please refer to FIG. 2' for a flowchart of a network telephone system connection according to an embodiment of the present invention. In step S200, the network phone host and the network phone are respectively turned on. Step S21G 'The network telephone handset uses the wireless communication protocol to group Wei-Shan information to: _ telephone domain. Go to S22Q, the slave management module of the network phone host picks up the registration information, and verifies the identity of the network phone handset to complete the connection and establish a wireless channel. Step S212, a user inputs an SIp account in the VoIP phone, and the account registration management module of the VoIP phone connects the sip account of the continuation to the sip account to send the login request to the VoIP host, requesting login (10) . Step S222 'The account management module of the network telephone host receives the login request ^, to complete the registration of the SIP account to the SIP server, establish a network connection between the network phone master and the SHM server, and connect to the network. The data path between the road connection and the wireless channel. Step S224, after the data channel is established, the VoIP host sends a successful login message to the VoIP handset. In step S214, the VoIP phone receives the message of successful login, and sends a synchronization request message to the VoIP host by the suffocating officer module, and the VoIP host synchronizes the contact and related setting information. Step ^ S226, the user management module of the VoIP host receives the message of the synchronization request, and completes the action of synchronizing the contact person and related setting information. When the synchronization is completed, the VoIP system is connected. ^ The foregoing process steps simply describe the way in which a network telephone system is connected, and only a single network telephone handset establishes a data path. Since the network telephone host of the network telephone of the present invention can simultaneously provide multiple SIP account logins, plural The principle of establishing a data path by a network telephone handset is the same as described above, so it will not be described here. Please refer to FIG. 3A, which is a flowchart of dialing a call for a network telephone system according to an embodiment of the present invention. When the VoIP system is successfully connected, the VoIP system can start making or receiving calls. Step S31〇, the user wants to make a call, the examiner wants to _ the other party's number, and the network telephone handset sends a call request to the network telephone host. In step S32, the VoIP host ^& receives the call request' and delivers the call request to the phantom server, waiting for the response from the SIP server. Step S33〇, the Slp server should pass back
話請求，並連線至受話方之網路電話主機。步驟S331，SIP 伺服器等待受話方應答。若受財有回應，接續步驟 幻32 SIP伺服器發送一成功應答至發話方的網路電話主 機端，若受話方無回應，則接續步驟S333，SIp伺服器等 待預疋時間後仍無回應，gp發送一應答失敗之訊息至發 話方的網路電話主機端。 β步驟S321，當發話方網路電話主機端接收成功應答之 A息，即傳遞至網路電話子機端。步•驟S311，網路電話子 機接收成魏答找息，縣賊立，_通話直到 任一方結束通話為止。 步驟S322，當發話方網路電話主機端·欠應答失敗之 訊心即傳遞至網路電話子機端。步驟如2，網路電話子 機接收應答失敗之訊息，通話通料被建立，將從營幕顯 示口知使用者’或是以鈴聲告知使用者應答失敗之訊息。 、明參考SUB’係為本發明-實細網路電話系統接 H話之/”也圖。當網路電話系統連結成功後，網路電話 系統即可’鑛或是接聽來電。轉從5，網路電話主 200847748 機端從SOM司服器接收一來電應答請求。步驟S326，網路 電話主機端根據來電應答請求之SIP帳戶，發出一接聽請 求至相對應之網路電話子機端。 〜 步驟S315，當使用者接受接聽請求，即接聽來電，網 路電話子機會發送一成功應答訊號至網路電話主機端。步 驟S327，網路電話主機接受網路電話子機之成功應答訊 號，將回應SIP伺服器的來電應答請求，建立通話通路， 開始通話直到任一方結束通話為止。 步驟S316，當使用者拒絕接受接聽請求，即拒絕接聽 來電，網路電話子機會發送一應答失敗訊號至網路電話主 機端。步驟S328 ,網路電話主機接受網路電話子機之應答 失敗訊號，將拒絕SIP伺服器的來電應答請求，通話通路 不被建立，此次的接聽請求並不成功。 ~ 前述兩流程步驟簡單說明一使用者撥號或接收一 SIP 通號之過程，由於本發明網路電話之網路電話主機可 同時提供多個SIP帳號登入，因此，本發明之網路電話主 機可在同一時間提供不同用戶，藉由上述之方式撥號或接 聽電話。 如先前技術中所述，習知之網路電話系統，往往僅支 持單一用戶的通訊協定帳號。本發明係揭露一可同時登錄 複數帳號之網路電話系統，使得每一個網路電話子機可註 冊一個網路電話帳號’透過網路電話主機與SIP伺服器達 成連結，提供使用者一更為便捷之通訊工具。值得注意的 13 200847748 是，在此實施例中，僅以SIP通訊協定作為本發明之實施 恶樣，然本發明並不欲於此限定本發明所揭露網路電話中 之通訊協定模式之種類或是數量，要應用於網路電話中其 餘之通訊協定模組，亦為熟悉該項技術者可依照本發明2 精神以及實施方式所輕易推知的。 5本發明所揭露之網路電話，即解決了前述之習知問 題，在同一部網路電話主機内可同時登錄多個帳號，只需 藉由複數個網路電話子機即可達成待機，在判 ^ 之随後’再蚊那—隻網路電話子機: 打後續之通話操作，成功地以巧思解決了網路 臨之設備繁狀問題，簡化了_請複數酬路電話主 機之設備數量。 本發明雖讀佳實__如上，然賤_以限定 本發明精神及發明實體。對所屬技術領域中具有通常知識 者’當可輕練解並其它元件衫絲產生相同的功 效。是以，在不麟树明之精神域圍崎作之修改， 均應包含在下述之申請專利範圍内。 【圖式簡單說明】 圖-係為本發1實__電話线之示意圖； •圖二係為本發明—實施_路電話系統連結之流程 圖， 200847748 圖三A係為本發明一實施例網路電話系統撥號請求通 話之流程圖；以及 圖三係為本發明一實施例網路電話系統接聽通話之 流程圖。 【主要元件符號說明】 1網路電話系統 10 SIP伺服器 . 12網路電話主機 14複數個網路電話子機 120帳號管理模組 122子機管理模組 124用戶管理模組 141第一子機 142第二子機 143第η子機 1410無線通信協議模組 1412帳號註冊管理模組 1414信息管理模組 S200〜S226各流程步驟 S310〜S333各流程步驟 15Request and connect to the recipient's Internet phone host. In step S331, the SIP server waits for the callee to answer. If there is a response, the SIP server sends a successful response to the VoIP terminal of the Talker. If the recipient does not respond, then proceed to step S333, and the SIp server waits for the preview time and does not respond. Gp sends a response failure message to the sender's network phone host. In step S321, when the calling party's network telephone host receives the A message of successful response, it is transmitted to the network telephone handset. Step S311, the VoIP phone receives the Wei answer, the county thief, _ call until either party ends the call. In step S322, when the sender's network telephone host has failed to respond, the message is transmitted to the network telephone handset. In step 2, the VoIP handset receives the message that the response failed, and the call material is established, and the user is notified from the screen display or the user is notified of the failure by the ringtone. The reference to SUB' is the invention - the actual fine network telephone system is connected to H / / also. When the VoIP system is successfully connected, the VoIP system can 'mine or answer the call. Transfer from 5 The VoIP host 200847748 receives an incoming call response request from the SOM server. In step S326, the VoIP host sends an answer request to the corresponding VoIP handset according to the SIP account of the incoming call response request. ~ Step S315, when the user accepts the answer request, that is, the incoming call is received, the network telephone sub-machine sends a successful response signal to the network telephone host. In step S327, the network telephone host accepts the successful response signal of the network telephone handset. In response to the SIP server's incoming call response request, the call path is established, and the call is started until either party ends the call. Step S316, when the user refuses to accept the answer request, the call is rejected, and the network caller sends a response failure signal to The network phone host side. In step S328, the network phone host accepts the response failure signal of the network phone handset, and will reject the incoming call of the SIP server. A request, the call path is not established, and the answer request is not successful. ~ The foregoing two process steps simply describe a process in which a user dials or receives a SIP number, because the network phone host of the network phone of the present invention can At the same time, multiple SIP account logins are provided. Therefore, the network telephone host of the present invention can provide different users at the same time, and dial or answer the call by the above manner. As described in the prior art, the conventional network telephone system often Only a single user's communication protocol account is supported. The present invention discloses a network telephone system that can simultaneously log in multiple accounts, so that each network telephone handset can register a network telephone account 'via the network telephone host and the SIP server. A link is provided to provide a more convenient communication tool for the user. It is noted that 13 200847748 is, in this embodiment, only the SIP communication protocol is used as the implementation of the present invention, but the present invention is not intended to be limited thereto. The invention discloses the type or quantity of the communication protocol mode in the network telephone, and applies the remaining communication in the network telephone. The protocol module can also be easily inferred by those skilled in the art according to the spirit and implementation of the present invention. 5 The network telephone disclosed in the present invention solves the aforementioned conventional problems in the same network telephone. In the host, multiple accounts can be logged in at the same time, and only a plurality of VoIP phones can be used to achieve standby. After the judgment, the 're-mosquito-only VoIP phone: the subsequent call operation succeeds. The problem is solved by the ingenuity of the network, and the number of devices of the multi-function telephone host is simplified. The present invention is better than the above, and then defines the spirit of the present invention and the inventive entity. For those who have the usual knowledge in the technical field, 'the same effect can be achieved if the other components are succinctly solved. Therefore, the revision of the spirit domain of the non-Lin Shuming is included in the following patent application scope. Inside. BRIEF DESCRIPTION OF THE DRAWINGS FIG. 2 is a schematic diagram of a telephone line connected to the present invention. FIG. 2 is a flow chart of the invention-implementation_road telephone system connection, 200847748 FIG. 3A is an embodiment of the present invention. A flow chart of a dialing request call by a network telephone system; and FIG. 3 is a flow chart of answering a call by the network telephone system according to an embodiment of the present invention. [Main component symbol description] 1 network telephone system 10 SIP server. 12 network telephone host 14 multiple network telephone sub-machine 120 account management module 122 sub-machine management module 124 user management module 141 first sub-machine 142 second slave 143 η slave 1410 wireless communication protocol module 1412 account registration management module 1414 information management module S200 ~ S226 process steps S310 ~ S333 process steps 15