201210265 六、發明說明： 【發明所屬之技術領域】 [0001] 本發明涉及一種網路電話運用技術，尤其涉及一種閘道 器及利用該閘道器建立網路電話通話的方法。 【先前技術·】 [0002] 隨著網路技術的不斷發展，用戶可藉由網路電話互相聯 繫。利用網路電話可有效幫助用戶節省通話費用。然而 ，在撥打網路電話前，用戶往往需要登陸電信公司的網 路平臺並且在本地用戶終端（例如，電腦等）下載軟體 電話（Soft Phone)，而網際網路用戶與日倶增，網路 經常會發生堵塞。如此，容易導致用戶很難被分配到足 夠的頻寬以撥打網路電話，並且網路電話的品質也會受 到嚴重的影響，例如，說話聲音斷斷續續。 [0003] 可見，如何保證網路電話的快速建立連接以及通話順暢 是非常重要並且亟需解決的問題，此外，如何避免在用 戶終端安裝過多的軟體也是提高用戶終端效能的方法。 【發明内容】 [0004] 鑒於以上内容，有必要提供一種閘道器及利用閘道器建 立網路電話通話的方法，可藉由訪問本地連接的閘道器 所提供的網頁直接撥打網路電話，無需在本地用戶終端 下載額外的軟體電話，從而方便用戶終端快捷撥打網路 電話並且藉由相關參數的設置保證通話品質。 [0005] 一種閘道器，該閘道器連接本地用戶終端以及藉由網路 連接電信公司的基地台，所述閘道器包括：設置模組， 用於設置針對網際網路語音協定VoIP封包的編碼方式， 099127823 表單編號A0101 第4頁/共21頁 0992048873-0 201210265 θ 以及δ又置針對不同編碼方式的服務品質QoS參數；對播敢 動協疋sip模組’用於在所述本地用戶終端撥打遠端用戶 、、端的VoI p電話後，啟動與所述的遠端用戶終端的s IP對 ^以雄疋所述本地用戶終端與所述遠端用戶終端之間 傳送VoIP封包的編碼方式；_模組，用於在所述sIp對 話成功完成後偵測從所述本地用戶終端傳送的ν〇ιρ封包 ’所述Voip封包具有即時傳輸協定RTp資訊·，參數解柝模 組，用於根據接收到的VoIP封包的編碼方式確定相應的 QoS參數，及根據所述閘道器與所述基地台之間網路的網 路協定解析所述Q〇S參數以生成請求封包;及通訊模錤， 用於將該請求封包發送至所述的基地台以請求建立與遠 端用戶終端的VoIP通話，並在所述請求成功後藉由所述 基地台發送所述VoIP封包至遠端用戶終端以建立v〇Ip通 話。 [0006] 一種利用閘道㈣立網路電話通話的方法，運用於間道 器中’該閘道器連接本地用戶終端以及藉由網路連接€ Ο 信公司的基地台，該方法包括：設置針對網際網路語音 協定VoIP封包的編碼方式，以及設置針對不同編碼方式 的服務品質QoS參數；在所述本地用戶終端撥打遠端用戶 終端的VoIP電話後，啟動與所述的遠，戶終端的sip對 話，以確定所述本地用戶終端與所述遠端用戶故端之Μ 傳送VoIP封包的編碼方式；在所述sip料 „ 對話成功完成後偵 測從所述本地用戶終端傳送的VoIP封台，^ ^所述VoIP封包 具有即時傳輸協定R T P資訊；根據接收到沾v m 的“❶封包的編 碼方式確定相應的QoS參數，及根據所$ 甲運器與所述基 099127823 表單編號A0101 第5頁/共21頁 0992048873-0 201210265 地口之間、周路的網路協定解析所述QoS參數以生成枝喪 包；將該請求封包發送至所述的基地台以請求建立:、/ 端用戶終端的VGlp通話；及在所述請求成功後藉由所= 基地口發送所述ν〇ΙΡ封包至遠端用戶终端以建立二 話。 14 [0007] [0008] [0009] 相較於習知技術，所述的間道器及利用閘道器建立網路 電話通話的方法’可藉由訪問本地連接❹ 1道器所提供 的網頁直接撥打網路電話，無需在本地好終端下載額 外的軟體電話’從而方便用戶終端快捷撥打網路電話並 且藉由相關參數的設置保證通話品質。 【實施方式】 如圖1所示，是本發明閘道器的較佳實施方式的運行環境 圖。本發明閘道器2用於協助本地用戶終端丨與遠端用戶 終端6建立通話連接以方便快捷的撥打網路電話。在本較 佳實施方式中，所述網路電話是指網際網路語音協定（201210265 VI. Description of the Invention: [Technical Field] [0001] The present invention relates to a network telephone application technology, and more particularly to a gateway and a method for establishing a network telephone call using the gateway. [Prior Art·] [0002] With the continuous development of network technology, users can connect with each other through Internet telephony. Using VoIP can help users save on calling costs. However, before making an Internet call, users often need to log in to the telecommunications company's network platform and download Softphones on local user terminals (for example, computers, etc.), while Internet users are increasing. Blockages often occur. As a result, it is easy for a user to be allocated a sufficient bandwidth to make an Internet call, and the quality of the Internet phone is also seriously affected, for example, the voice is intermittent. [0003] It can be seen that how to ensure the fast connection establishment and smooth communication of the network telephone is very important and urgently needed to be solved. In addition, how to avoid installing too many softwares in the user terminal is also a method for improving the performance of the user terminal. SUMMARY OF THE INVENTION [0004] In view of the above, it is necessary to provide a gateway device and a method for establishing a network telephone call by using a gateway device, which can directly make an Internet call by accessing a webpage provided by a locally connected gateway device. There is no need to download an additional software phone at the local user terminal, so that the user terminal can quickly make a network call and ensure the call quality by setting related parameters. [0005] A gateway device that connects a local user terminal and a base station of a telecommunications company via a network, the gateway device comprising: a setting module, configured to set a VoIP packet for an Internet voice protocol Encoding method, 099127823 Form number A0101 Page 4/Total 21 page 0992048873-0 201210265 θ and δ are set to service quality QoS parameters for different coding methods; for the broadcast sip module sip module 'for local After the user terminal dials the remote user's VoI p phone, the s IP pair with the remote user terminal is activated to transmit the code of the VoIP packet between the local user terminal and the remote user terminal. a module for detecting a ν〇ιρ packet transmitted from the local user terminal after the sIp session is successfully completed. The Voip packet has an instant transfer protocol RTp information, and the parameter decoding module is configured according to Decoding the received VoIP packet to determine a corresponding QoS parameter, and parsing the Q〇S parameter according to a network protocol of the network between the gateway and the base station to generate a request packet; And a communication module, configured to send the request packet to the base station to request to establish a VoIP call with the remote user terminal, and send the VoIP packet to the remote by the base station after the request is successful The end user terminal establishes a v〇Ip call. [0006] A method for using a gateway (four) to establish a telephone conversation in a gateway, wherein the gateway is connected to a local user terminal and connected to a base station of the company by a network, the method includes: setting A method for encoding the VoIP packet of the Internet voice protocol, and setting a quality of service QoS parameter for different coding modes; after the local user terminal dials the VoIP phone of the remote user terminal, starting with the remote terminal a sip session to determine a coding mode for transmitting the VoIP packet between the local user terminal and the remote user; detecting the VoIP closure transmitted from the local user terminal after the sip material is successfully completed , ^ ^ The VoIP packet has an instant transfer protocol RTP information; the corresponding QoS parameter is determined according to the encoding method of the "❶ packet received by the vm, and according to the $0129, the form number A0101, page 5 / 21 pages 0992048873-0 201210265 The network protocol between the ground and the perimeter resolves the QoS parameters to generate a packet; the request packet is sent to the The base station establishes: VGlp call of the end user terminal; and/or after the request is successful, the ν〇ΙΡ packet is sent to the remote user terminal by the base port to establish a second call. [0009] [0009] [0009] Compared with the prior art, the inter-channel device and the method for establishing a network telephone call by using a gateway can be accessed by accessing a webpage provided by a local connection device. Directly dial the Internet phone, no need to download additional software phone in the local good terminal', so that the user terminal can quickly make Internet calls and ensure the call quality through the setting of relevant parameters. [Embodiment] As shown in Fig. 1, it is an operating environment diagram of a preferred embodiment of the gateway of the present invention. The gateway device 2 of the present invention is used to assist the local user terminal to establish a call connection with the remote user terminal 6 to conveniently and quickly make a network call. In the preferred embodiment, the Internet telephony refers to an internet voice protocol (
Voice Over Internet Prot〇c〇l ’ VoIP)電話，下 文以V ο IP電話為例進行說明；。： 所述本地用戶終端1藉由所述閘道器2發送資料封包至電 信公司的基地台3以請求建立與遠端用戶終端6的通話連 接。所述基地台3藉由網路與電信公司的伺服器4相連接 ，所述伺服器4可處理所述基地台3接收到的資料封包， 並藉由所述基地台3發送回應封包至所述的閘道器2。然 後，所述閘道器2根據回應封包的内容藉由所述基地台3 傳送相關語音封包至所述遠端用戶終端6以實現與遠端用 戶終端6的通話。 099127823 表單煸號Α0101 第6頁/共21頁 0992048873-0 201210265 [0010] [0011] ο [0012] [0013] Ο [0014] 所述本地用戶終端1與遠端用戶終端6是相對而言，同理 ，所述遠端用戶終端6也可藉由相連的閘道器5主動建立 與所述本地用戶終端1的VoIP通話。所述本地用户終端j 與遞細用戶終可以是電腦、手機、個人數位助理等電 子裝置。 所述閘道器2與所述基地台3之間的網路連接可以是全球 立通微波存取（Worldwide Interoperability f〇r Microwave Access, WiMAX)網路、2G (SecondGeneration)或3G (Third'Generation)網路。 .....’ ：.. 所述本地用，戶終端1包括的VoIP單元10以及所述遠端用戶 終端6包括VolP單元60可以藉由所述閘道器2提供的網路 電話網頁進行下載，下文將作詳細說明》所述v〇lp單元 10、60用於根據語音資訊生成v〇IP封包。藉由傳送所述 VoIP封包可實現所述本地用戶終端1與遠端用戶終端6之 間的VoIPii話。 如圖2所示’是本發明閘道器的較佳實施方式的功能辦組 圖。在本較佳實施方式中，所述的閘道器2還包括處理器 20與儲存裝置21。所述處理器20用於執行所述閘道器2中 安裝或嵌入的各類軟體。所述儲存裝置21用於儲存各類 資料，例如通訊錄、所設置的各類參數等資料。 在本較佳實施方式中，所述的閘道器2還包括多個功能模 組’分別是：設置模組22、登錄模組23、對話啟動協定 (Session Initiation Protocol，SIP)模組24、 偵測模組25、參數解析模組26以及通訊模組27。 099127823 表單編號A0101 第7頁/共21頁 0992048873-0 201210265 [0015] 所述的設置模組2 2，用於在所述閘道器2設置針對v〇 i p封 包的編碼方式（Codec)，並設置針對不同編碍方式的服 務品質（Quality of Service，Q〇s)參數，以及將上 述設置的資料儲存至所述的儲存裝置21中。 [0016] 例如’ VoIP封包編碼方式可以是G729、G723、PCMU、 PCMA、G726-32等。QoS參數是一種控制機制，它提供了 針對不同用戶或者不同資料流程採用相應不同的優先順 序’或者是根據應用程式的要求，保證資料流程的性能 達到一定的水準。q〇S參數可使得所述串流多媒體應用保 持固定的傳輪率，並減少延時。所設置的q〇S參數可根據 實際需求進行修改。 [0017] 所述的登錄模組23，用於在所述閘道器2啟動後登錄VoIP 帳號。若所述閘道器2尚未有VoIP帳號，所述登錄模組23 可自動向電信公司的伺服器4註冊VoIP帳號。 [0018] 所述的登錄模組23，還用於為衝;逮本地用戶終端1提供網 路電話網頁’所述網路電話網頁可顯示通訊錄、VoIP單 元10的插件以及撥號（Dial )按鈕。其中，所述通訊錄 列舉了聯繫人的各項資訊，例如，聯繫人所使用的遠端 用戶終端6的位址、Vo IP電話號碼等資訊。 [0019] 所述登錄模組23，進一步用於在所述本地用戶終端1開啟 所述網路電話網頁時判斷所述本地用戶終端1是否安裝有 V〇IP單元1〇 ’並在所述本地用戶終端^支有安裝VoIP單 元10時下載所述VoIP單 元10的插件至所述本地用戶終端 1以使得所述V〇lp單元1〇能夠根據用戶的語音資訊生成 099127823 表單編號ΑΟίοι 第8頁/共2丨頁 0992048873-0 201210265 [0020] [0021]Ο [0022] [0023]❹ [0024] ν〇ΐρ封包《所述ν〇ΙΡ單元10的插件可以是應用程式段。 藉由所述的網路電話網頁，用戶藉由所述本地用戶終端1 撥打網路電話時，無需安裝軟體電話（SoftPhone)， 只需訪問所述閘道器2提供的網路電話網頁，藉由下載 VoIP單元10的插件至本地用戶終端1，即可實現撥打網路 電話，並且後續無需再次下載即玎運用所述的化1?單元 10 ° 開啟所述閘道器2的電源使所述閘道器2啟動後，用戶即 可瀏覽所述網路電話網冥查找聯繫人，在確認聯繫人後 v/ 可點擊撥號按鈕。 所述的SIP模組24，用於在所述本地用戶終端1撥打遠端 用戶終端6的VoIP電話後’啟動與所述的遠端用戶終端6 的SIP對話，以確定所述本地用戶終端1..與所述遠端用戶 終端6之間傳送v〇 I p封包的編瑪方式。 如上所述’ VoIP封包編碼方式可以是G729、G723、 PCMU、PCMA、G726-32等》發起對話的主叫方（例如， 本地用戶終端1 )與被叫方（例如’遠端用戶終端6)所 支援的VoIP封包的編瑪方式可能是不相同的。例如，所 述本地用戶終端1支援的編碼方式是G729、G72^pcmu ，而所述遠端用戶終端6支援的编庇+上β 碼碼方式是PCMU、PCMA 及G726-32。因此’藉由上述的叫對話可確定主叫方 與被叫方共同支持的VoIP封包的編瑪方式從而保證 VoIP通話的進行。 上述的S1P對話可以包括如下的過程：主叫方獲得被叫方 099127823 表單編號A0101 第9頁/共21頁 0992048873-0 201210265 的位址’該位址可以表達為“用戶名@網域名稱”；藉由 網域名稱系統（DNS，Domain Name System)將被叫用 戶位址轉化為網際網路協定（IP，lnternet Pr〇t〇c〇1 )地址’從而啟動SIP呼叫信令過程；主叫方根據被叫方 的IP位址向其發送SIP INVITE請求；被叫方應答，沿原 路返回SIP 200 0K的消息；主叫方沿原路徑回送ACK消 息，成功完成SIP對話。 [0025] [0026] [0027] 在所述SIP對話成功完成後’所述本地用戶終端1的v〇Ip 單元10根據用戶的語音資訊生成VoIP封包，並將該v〇Ip 封包傳送至所述閘道器2。 所述的偵測模組25，用於偵測從所述本地用戶終端1傳送 的VoIP封包。所述VoIP封包具有即時傳輸協定 (Real-Time Transport Protocol，RTP)資訊，該 RTP資訊包含時間戳等資料。 所述的參數解析模組26 ’用於板棣接收到的VoIP封包的 編碼方式確定相應的qos參數。树茹；，q〇S的參數中包括 封包傳送的時間間隔（可用..P t i m,e ”表示）、每個封包 的位數（bits)、每個時間間隔内傳送的位數（可用“ vif表示）、標頭長度（hea(jer_ien)、頻寬（ Bandwidth)等數據。利用所述的q〇s參數可保證通話品 質。 假5又所述¥〇1?封包的編碼方式是^腳時’卩>(;丨11]6=2〇1115 ’表示每20ms傳送一個封包；每個封包的長度為 bits;vif = Ptime * 64 = 128〇’ 表示每2〇ms可以 099127823 表單編號A0101 第10頁/共21頁 0992048873-0 [0028] 201210265 傳送1280 bits ;頻寬可用如下公式進行計算： [0029] Band- width=8*(vif/8+header_len)*(i〇〇〇/ptime)=8*(l 280/8+40)*50=80000 (bits/Sec)。 [0030] 所述的參數解析模組26，還用於根據所述閘道器2與所述 基地台3之間網路的網路協定解析所述q〇s參數以生成請 求封包。例如，WiMAX網路的網路協定是WiMAX協定，3G 網路的網路協定可包括WCDMA及CDMA2000。 [0031] 所述的通訊模組27 ’甩於將該請求封包發送至所述的基 地台3以請求建立與遠端用戶終端6的¥〇1?通話，並從所 述基地台3接收回應封包。 f [0032] 所述的通訊模組27可根據所述回應封包確認所述建立與 遠端用戶終端6的VoIP通話的請求是否成功。例如，該請 求封包中包括對頻寬的要求，藉由所述回應封包可確定 所述基地台3是否能夠分配所要求的頻寬給所述的本地用 戶終端1實現與所述遂端用戶終端6的v〇ip通話。 [0033] 所述的通訊模組27，還用於在所述請求成功後將所述 Vo IP封包中的KTP資訊放入增強的即時輪詢服務 (Extended real-time Polling Service > ertPS) 佇列中，並從所述基地台3下載上傳排程（UpHnk Map )，並根據所述上傳排程藉由所述基地台3發送所述π 封包至遠端用戶終端6以建立VoIP通話。 [0034]所述上傳排程中可列舉所述基地台3對不同的用戶終端發 出的VoIP通話請求的排配。根據該上傳排程，所述通訊 0992048873-0 丨表單編號A0101 第11頁/共21頁 201210265 &組27即可依據確定的時間或次序傳送所述VqIP封包至 遠端用戶終端6。 [] h圖3所不’是本發明利用閉道器建立網路電話通話的方 法的較佳實施方式的流程圖。首先，步驟S2 ’所述的設 置权組2 2在所述閘道器2設置針對ν〇ϊρ封包的編碼方式， 並设置針對不同編碼方式的Q〇s參數，以及將上述設置的 資料儲存至所述的儲存裝置21中。 _6]步驟S4 ’所述的登錄模組23在所述開道器2啟動後登錄Voice Over Internet Prot〇c〇l ’ VoIP) phone, the following is an example of a V ο IP phone; The local user terminal 1 transmits a data packet to the base station 3 of the telecommunications company by the gateway 2 to request to establish a call connection with the remote user terminal 6. The base station 3 is connected to the server 4 of the telecommunication company via a network, and the server 4 can process the data packet received by the base station 3, and send the response packet to the base station 3 by the base station 3. The gateway 2 described. Then, the gateway 2 transmits a relevant voice packet to the remote user terminal 6 by the base station 3 according to the content of the response packet to implement a call with the remote user terminal 6. 099127823 Form 煸 Α 0101 Page 6 / Total 21 Page 0992048873-0 201210265 [0011] [0012] [0014] [0014] The local user terminal 1 and the remote user terminal 6 are relatively speaking, Similarly, the remote user terminal 6 can also actively establish a VoIP call with the local user terminal 1 by the connected gateway 5. The local user terminal j and the delegating user may end up being an electronic device such as a computer, a mobile phone, or a personal digital assistant. The network connection between the gateway 2 and the base station 3 may be a Worldwide Interoperability f〇r Microwave Access (WiMAX) network, 2G (Second Generation) or 3G (Third'Generation). )network. .....': The local use, the VoIP unit 10 included in the household terminal 1 and the remote user terminal 6 including the VolP unit 60 can be performed by the web phone webpage provided by the gateway 2. The download will be described in detail below. The v〇lp unit 10, 60 is used to generate a v〇IP packet based on the voice information. The VoIP ii between the local user terminal 1 and the remote user terminal 6 can be implemented by transmitting the VoIP packet. As shown in Fig. 2, ' is a functional group diagram of a preferred embodiment of the gateway of the present invention. In the preferred embodiment, the gateway 2 further includes a processor 20 and a storage device 21. The processor 20 is configured to execute various types of software installed or embedded in the gateway 2. The storage device 21 is configured to store various types of materials, such as an address book, various parameters set, and the like. In the preferred embodiment, the gateway device 2 further includes a plurality of function modules, namely: a setting module 22, a login module 23, and a Session Initiation Protocol (SIP) module 24, The detection module 25, the parameter analysis module 26, and the communication module 27. 099127823 Form No. A0101 Page 7 / 21 page 0992048873-0 201210265 [0015] The setting module 2 2 is configured to set a coding mode (Codec) for the v〇ip packet in the gateway 2, and A quality of service (Q〇s) parameter for different ways of arranging is set, and the above set data is stored in the storage device 21. [0016] For example, the VoIP packet coding method may be G729, G723, PCMU, PCMA, G726-32, or the like. The QoS parameter is a control mechanism that provides a different priority order for different users or different data flows or ensures that the performance of the data flow reaches a certain level according to the requirements of the application. The q〇S parameter allows the streaming multimedia application to maintain a fixed pass rate and reduce latency. The set q〇S parameters can be modified according to actual needs. [0017] The login module 23 is configured to log in to the VoIP account after the gateway 2 is activated. If the gateway 2 does not have a VoIP account, the login module 23 can automatically register a VoIP account with the server 4 of the telecommunications company. [0018] The login module 23 is further configured to provide a network phone webpage for the local user terminal 1. The network phone webpage can display an address book, a plug-in of the VoIP unit 10, and a dial button. . The address book lists various information of the contact, for example, the address of the remote user terminal 6 used by the contact, the Vo IP phone number and the like. [0019] The login module 23 is further configured to determine, when the local user terminal 1 opens the webpage webpage, whether the local user terminal 1 is installed with a V〇IP unit 1〇′ and is in the local The user terminal supports downloading the plug-in of the VoIP unit 10 to the local user terminal 1 when the VoIP unit 10 is installed, so that the V〇lp unit 1 can generate 099127823 according to the voice information of the user. Form number ΑΟίοι Page 8/ A total of 2 pages 0992048873-0 201210265 [0020] [0023] [0023] [0024] ν〇ΐρPacket The plug-in of the ν〇ΙΡ unit 10 may be an application segment. By using the VoIP phone webpage, when the user dials the VoIP call by the local user terminal 1, the user does not need to install the softphone (SoftPhone), and only needs to access the VoIP phone webpage provided by the gateway device 2, By downloading the plug-in of the VoIP unit 10 to the local user terminal 1, the network call can be made, and the power of the gateway 2 can be turned on by using the said 1? unit 10 ° without the need to download again. After the gateway 2 is activated, the user can browse the VoIP network to find a contact, and after confirming the contact, v/ can click the dial button. The SIP module 24 is configured to: initiate a SIP conversation with the remote user terminal 6 after the local user terminal 1 dials the VoIP phone of the remote user terminal 6 to determine the local user terminal 1 The arranging mode of transmitting the v〇I p packet with the remote user terminal 6. As described above, the VoIP packet coding method may be G729, G723, PCMU, PCMA, G726-32, etc., the calling party (for example, the local user terminal 1) and the called party (for example, the 'remote user terminal 6'). The encoding method of the supported VoIP packets may be different. For example, the encoding mode supported by the local user terminal 1 is G729, G72^pcmu, and the method of scribing + upper beta code supported by the remote user terminal 6 is PCMU, PCMA and G726-32. Therefore, by means of the above-mentioned called dialogue, the naming mode of the VoIP packet supported by the calling party and the called party can be determined to ensure the progress of the VoIP call. The above S1P dialog may include the following process: the calling party obtains the called party 099127823 Form number A0101 Page 9 / 21 pages 0992048873-0 201210265 address 'This address can be expressed as "user name@domain name" The domain name system (DNS, Domain Name System) is used to translate the called user address into an Internet Protocol (IP, Ethernet Pr〇t〇c〇1) address to initiate the SIP call signaling process; The party sends a SIP INVITE request to the called party according to the IP address of the called party; the called party replies and returns the SIP 200 0K message along the original path; the calling party sends an ACK message along the original path to successfully complete the SIP conversation. [0027] After the SIP dialog is successfully completed, the v〇Ip unit 10 of the local user terminal 1 generates a VoIP packet according to the voice information of the user, and transmits the v〇Ip packet to the Gateway 2 The detecting module 25 is configured to detect a VoIP packet transmitted from the local user terminal 1. The VoIP packet has Real-Time Transport Protocol (RTP) information, and the RTP information includes data such as a time stamp. The parameter parsing module 26' is configured to determine the corresponding qos parameter for the encoding mode of the VoIP packet received by the board. Shuru;, q〇S parameters include the time interval of packet transmission (represented by ..P tim, e), the number of bits per packet (bits), the number of bits transmitted in each time interval (available " Vif indicates), header length (hea (jer_ien), bandwidth (Bandwidth) and other data. The q〇s parameter can be used to guarantee the call quality. False 5 is also described as ¥〇1? The encoding method of the packet is ^foot When '卩>(;丨11]6=2〇1115' means that one packet is transmitted every 20ms; the length of each packet is bits; vif = Ptime * 64 = 128〇' means that every 2〇ms can be 099127823 Form No. A0101 Page 10 of 21 0992048873-0 [0028] 201210265 transmits 1280 bits; the bandwidth can be calculated by the following formula: [0029] Band-width=8*(vif/8+header_len)*(i〇〇〇/ptime = 8 * (l 280 / 8 + 40) * 50 = 80000 (bits / Sec) [0030] The parameter analysis module 26 is further configured to use the gateway 2 and the base station 3 The network protocol between the networks resolves the parameters of the q〇s to generate a request packet. For example, the network protocol of the WiMAX network is a WiMAX protocol, and the network protocol of the 3G network may include WCDMA and CDMA. [0031] The communication module 27' sends the request packet to the base station 3 to request to establish a call with the remote user terminal 6, and from the base station 3 Receiving a response packet. [0032] The communication module 27 may confirm, according to the response packet, whether the request for establishing a VoIP call with the remote user terminal 6 is successful. For example, the request packet includes a bandwidth. It is required that the response packet can determine whether the base station 3 can allocate the required bandwidth to the local user terminal 1 to implement a v〇ip call with the terminal user terminal 6. [0033] The communication module 27 is further configured to put the KTP information in the Vo IP packet into an Extended Real-time Polling Service > ertPS queue after the request is successful, and Downloading an upload schedule (UpHnk Map) from the base station 3, and transmitting the π packet to the remote user terminal 6 by the base station 3 according to the upload schedule to establish a VoIP call. The upload schedule can enumerate the base station 3 for different user ends. The allocation of the VoIP call request issued by the terminal. According to the upload schedule, the communication 0992048873-0 丨form number A0101 page 11 / 21 page 201210265 & group 27 can transmit the VqIP according to the determined time or order The packet is packetized to the remote user terminal 6. [0] Figure 3 is a flow diagram of a preferred embodiment of the method of establishing a network telephone call using a closed channel. First, the setting right group 2 2 described in step S2' sets the encoding mode for the ν〇ϊρ packet in the gateway device 2, sets the Q〇s parameter for different encoding modes, and stores the above set data to In the storage device 21 described. _6] The login module 23 described in step S4' is logged in after the opener 2 is started.
Vo IP帳號’並為所述本地用戶終端丨提供網路電話網頁。 所述網路電話網頁可顯示撥號按鈕以及列桊聯繫人資訊 的通訊錄。 [0037] 步驟S6 ’所述的sip模組24在所述本地用戶終端丨撥打遠 端用戶終端6的VoIP電話後，啟動與所述的遠端用戶終端 6的SIP對話’以確定所述本地用戶終端1與所述遠端用戶 終端6之間傳送VoIP封包的碥_方式。 [0038] 在所述SIP對話成功完成後，所述丰地用戶終端1的VoIP 單元10根據用戶的語音資訊生成VoIP封包，並將該VoIP 封包傳送至所述閘道器2。 [0039] 步驟S8，所述的偵測模組25偵測從所述本地用戶終端1傳 送的VoIP封包。所述v〇IP封包具有RTP資訊，該RTP資訊 包含時間戳等資料。 [0040] 步驟S10，所述的參數解析模組26根據接收到的VoIP封 包的編碼方式確定相應的q〇S參數’並根據所述閘道器2 與所述基地台3之間網路的網路協定解析所述QoS參數以 099127823 表單編號A0101 第12頁/共21買 0992048873-0 201210265 [0041] [0042] [0043] Ο [0044] [0045] [0046] Ο [0047] [0048] [0049] 生成請求封包。 步驟S12，所述的通訊模組27將該請求封包發送至所述的 基地台3以請求建立與遠端用戶終端6的ν〇ΐρ通話，並從 所述基地台3接收回應封包。 步驟S14 ’所述的通訊模組27根據所述回應封包轉認所述 建立與遠端用戶終端6的VoIP通話的請求是否成功。 若所述請求不成功’流程返回至步驟S12，繼續發送該請 求封包至所述基地台3。 — .. ..... 若所述請求成功，於步驟S1&，所述的通訊模組27將所述 VoIP封包中的RTP資訊放入ertPS佇列中》 步驟S18，所述的通訊模組27從所述基地台3下載上傳排 程（Up1 ink Map)。 步驟S20，所述的通訊模組27根據所述上傳排程藉由所述 基地台3發送所述VoIP封包至遠端用戶终端6以建立v〇Ip 通話，然後，結束本流程。 綜上所述’本發明符合發明專利要件，爰依法提出專利 申請。惟，以上所述者僅為本發明之較佳實施例，本發 明之範圍並不以上述實施例為限，舉凡熟悉本案技藝之 人士援依本發明之精神所作之等效修飾或變化，皆應涵 蓋於以下申請專利範圍内。 【圖式簡單說明】 圖1是本發明閘道器的較佳實施方式的運行環境圖。 圖2疋本發明閑道器的較佳實施方式的功能模組圖。 099127823 表單編號A0101 第丨3頁/共21頁 0992048873-0 201210265 [0050] 圖3是本發明利用閘道器建立網路電話通話的方法的較佳 實施方式的流程圖。 【主要元件符號說明】 [0051] 本地用戶終端：1 [0052] VoIP單元：10、60 [0053] 閘道器：2、5 [0054] 處理器：2 0 [0055] 儲存裝置：21 [0056] 設置模組：22 [0057] 登錄模組：23 [0058] 對話啟動協定模組：24 [0059] 偵測模組：25 [0060] 參數解析模組：26 [0061] 通訊模組：27 [0062] 基地台：3 [0063] 伺服器：4 [0064] 遠端用戶終端：6 0992048873-0 099127823 表單編號A0101 第14頁/共21頁The Vo IP account' provides a web phone webpage for the local user terminal. The web phone web page can display a dial button and an address book listing the contact information. [0037] The sip module 24 described in step S6' initiates a SIP conversation with the remote user terminal 6 after the local user terminal dials the VoIP phone of the remote user terminal 6 to determine the local The 碥_mode of transmitting the VoIP packet between the user terminal 1 and the remote user terminal 6. [0038] After the SIP dialog is successfully completed, the VoIP unit 10 of the rich user terminal 1 generates a VoIP packet according to the user's voice information, and transmits the VoIP packet to the gateway 2. [0039] Step S8, the detecting module 25 detects a VoIP packet transmitted from the local user terminal 1. The v〇IP packet has RTP information, and the RTP information includes data such as a timestamp. [0040] Step S10, the parameter parsing module 26 determines a corresponding q〇S parameter ' according to the encoded manner of the received VoIP packet and according to the network between the gateway 2 and the base station 3 The network protocol parses the QoS parameters to 099127823 Form No. A0101 Page 12/Total 21 Buy 0992048873-0 201210265 [0042] [0044] [0044] [0046] [0048] [0048] [0049] Generating a request packet. In step S12, the communication module 27 sends the request packet to the base station 3 to request to establish a ν〇ΐρ call with the remote user terminal 6, and receives a response packet from the base station 3. The communication module 27 of the step S14 ′ forwards whether the request for establishing a VoIP call with the remote user terminal 6 is successful according to the response packet. If the request is unsuccessful, the flow returns to step S12 to continue transmitting the request packet to the base station 3. - .. ..... If the request is successful, in step S1&, the communication module 27 puts the RTP information in the VoIP packet into the ertPS queue, step S18, the communication mode Group 27 downloads an Up1 ink map from the base station 3. In step S20, the communication module 27 sends the VoIP packet to the remote user terminal 6 by the base station 3 according to the upload schedule to establish a v〇Ip call, and then ends the process. In summary, the invention conforms to the patent requirements of the invention, and the patent application is filed according to law. The above is only the preferred embodiment of the present invention, and the scope of the present invention is not limited to the above-described embodiments, and equivalent modifications or variations made by those skilled in the art in light of the spirit of the present invention are It should be covered by the following patent application. BRIEF DESCRIPTION OF THE DRAWINGS Fig. 1 is a diagram showing the operating environment of a preferred embodiment of the gateway of the present invention. 2 is a functional block diagram of a preferred embodiment of the tracker of the present invention. 099127823 Form No. A0101 Page 3 of 21 0992048873-0 201210265 [0050] FIG. 3 is a flow chart of a preferred embodiment of a method for establishing a network telephone call using a gateway. [Main component symbol description] [0051] Local user terminal: 1 [0052] VoIP unit: 10, 60 [0053] Gateway: 2, 5 [0054] Processor: 2 0 [0055] Storage device: 21 [0056] ] Setting Module: 22 [0057] Login Module: 23 [0058] Dialogue Startup Protocol Module: 24 [0059] Detection Module: 25 [0060] Parameter Analysis Module: 26 [0061] Communication Module: 27 [0062] Base station: 3 [0063] Server: 4 [0064] Remote user terminal: 6 0992048873-0 099127823 Form number A0101 Page 14 of 21