TW560163B - Method for dialing Internet protocol phone - Google Patents
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560163 五、發明說明(1) 【發明領域】 本發明是有關於一種撥打網路電話(Internet protocol phone,IP phone/1-Phone )之方法,且特別是 有關於一種利用整合封包無線電服務(general packet radio service,GPRS)網路來傳輸以撥打網路電話之方 法0 【發明背景】 由於電信科技的進步,使得人們雖位於不同地點,也 可藉由電話來進行溝通。而電話的連接係經由公眾交換電 話網路(public switched telephone network,PSTN) 之系統來建立。雖然電話的使用改善了人們的溝通方式, 彷彿縮短人們間的距離;但其收費卻不便宜,尤其是長途 電話甚至是國際電話之計費。 也因此,網路電話(lnternet pr〇t〇c〇1 ph〇ne , IP pone/ I -Phone )之技術因應而現世。網路電話係以網際網 路(Internet)或電腦網路取代部份或全部的pstn,其主 要技術是將語音數位化後,經過壓縮,再以丨p封包 (packet )形態於網際網路或電腦網路中傳輸至收話方。 收話方再將IP封包解壓縮並還原為語音,藉此執行通話功 能。如此,網路電話既能達到兩端溝通對話的目的,更町 減少通話費用。 而網路電話依通訊方式之不同,可區為三種,分別是 電腦對電腦(PC-to-PC)、電腦對電話(pc —1〇 —ph〇ne)560163 V. Description of the Invention (1) [Field of the Invention] The present invention relates to a method for dialing an Internet protocol phone (IP phone / 1-Phone), and in particular to a method for utilizing integrated packet radio services (general Packet radio service (GPRS) network transmission method to make Internet calls 0 [Background of the Invention] Due to the advancement of telecommunications technology, people can communicate by telephone even though they are located in different locations. The telephone connection is established via a public switched telephone network (PSTN) system. Although the use of telephones improves the way people communicate, it seems to shorten the distance between people; however, the charges are not cheap, especially for long-distance calls and even international calls. Because of this, the technology of Internet telephone (internet pr0toc1 phone, IP pone / I-Phone) came into being. Internet calling is to replace some or all of the pstn with the Internet or computer network. The main technology is to digitize the voice, compress it, and then use the packet format (p) on the Internet or Computer network to the receiver. The recipient then decompresses the IP packet and restores it to voice to perform the call function. In this way, Internet telephony can both achieve the purpose of communication and dialogue at both ends, and reduce call costs. According to the different communication methods, there are three types of Internet phones: computer-to-PC (PC-to-PC) and computer-to-phone (pc —10—ph〇ne).
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及電話對電話(Phone-to-Phone)。請參考第1圖,其所 繪示為網路電話所運用之系統與網路的示意圖。於第i圖 中,電腦1 0 1與電腦1 0 5係分別經由網際網路服務提供者 (Internet service provider,ISP)所提供之飼服器 1 0 3與伺服器1 〇 7連接上網際網路。而電話111與電話丨2 1八 別連接PSTN 1 13與PSTN 123。 乃 如第1圖中所示,若電腦101欲與電腦1〇5進行網路電 話之連接,電話101需先撥打一 IP電話號碼(譬如是1?位 址、或特定ID ),再經由伺服器1 〇 3、網際網路1 〇 〇而與電 腦1 0 5連結,以執行網路電話功能。若是電腦1 〇 1欲以網路 電話方式連接電話111,電腦101需先撥打一 ISP訂定之附 加碼及電話111之電話號碼,再經由伺服器1 〇 3、網際網路 1 0 0連接到網際網路電話服務提供者(I n t e r n e t telephony service provider,ITSP )所提供的網際網路 電話閘道器(Internet telephony gateway,I)ii5 而ITG 11 5係用以做語音訊息與IP封包之間的轉換。故, 此ITG 115係將電腦101所傳送之IP封包轉換成語音訊声, 再經由PSTN 113傳送給電話111。如此的方式可使得電"腦’ 1 0 1與電話111進行對話。而電話11 1與電話1 2 1間的網路電 話連結原理亦如同上述。 雖然網路電話之費用已較單純使用PSTN之電話費用便 宜許多,但在使用上得先撥打一 I p電話號碼,再經由饲服 器來進行連線’步驟相當繁瑣。且更因現今日漸普遍之行 動通訊裝置(如行動電話)尚無法撥打網路電話,如此董子And phone-to-phone. Please refer to Figure 1, which shows a schematic diagram of the system and network used by the Internet phone. In Fig. I, computer 101 and computer 105 are connected to the Internet via server 103 and server 107 provided by an Internet service provider (ISP), respectively. road. On the other hand, telephone 111 and telephone 丨 2 1 8 do not connect PSTN 1 13 and PSTN 123. As shown in the first figure, if the computer 101 wants to make a network telephone connection with the computer 105, the telephone 101 needs to dial an IP phone number (such as a 1? Address, or a specific ID), and then the server Device 103, Internet 100, and computer 105 to perform Internet phone functions. If the computer 1 〇1 wants to connect to the phone 111 by Internet phone, the computer 101 needs to dial an additional code set by the ISP and the phone number of the phone 111, and then connect to the Internet through the server 1 〇 03, the Internet 1 0 0 Internet telephony gateway (I) ii5 provided by Internet telephony service provider (ITSP) ii5 and ITG 11 5 is used to convert between voice messages and IP packets . Therefore, the ITG 115 converts the IP packets transmitted by the computer 101 into voice signals, and then transmits them to the phone 111 via the PSTN 113. In this way, a conversation can be made between the "brain" 1 01 and the telephone 111. The principle of the network telephone connection between telephone 11 1 and telephone 1 2 1 is the same as described above. Although the cost of Internet telephony is much cheaper than that of using PSTN alone, in order to use it, it is necessary to dial an IP phone number and then connect through the feeder. The steps are quite tedious. And because mobile communication devices (such as mobile phones) that are becoming more common today are not yet able to make Internet calls,
560163 五、發明說明(3) 使用行動通訊裝置之使用者而言,並無法降低使用者之通 訊費用。 【發明目的及概述】 有鑑於此,本發明的目的就是在提供一種利用GPRS網 路以撥打網路電話之方法。本發明之方法係供行動通訊裝 置之使用者可簡易地撥打網路電話,並能減少通話費用。 根據本發明的目的,提出一種撥打網路電話之方法, 此方法係用以提供一主叫方藉由一 GPRS網路撥打網路電話 給一被叫方,其中,該主叫方與該被叫方分別具有一主叫 方IP位址及一被叫方IP位址。此方法首先開始於主叫方發 送一要求通話訊息至被叫方;其中,此要求通話訊息包括 一主叫方IP位址以及一主叫方通訊端口(port )。接著, 被叫方接收此要求通話訊息,而得知主叫方IP位址以及主 叫方通訊端口。爾後,被叫方將其使用者語音轉碼成一第 一語音數據流,並將此第一語音數據流壓縮打包(pack ) 成至少一使用者資料包協定(user da t agr am pr ot oco 1, UDP )封包,其中此UDP封包係具有被叫方ip位址。接著, 被叫方依據主叫方IP位址及主叫方通訊端口且藉由⑶以網 路傳送此UDP封包至主叫方。再由主叫方由其主叫方通訊 端口接收此UDP封包,並進行解開(unpack )、解壓縮的 動作而成一第一語音數據流。此第二語音數據流再被解碼 成使用者語音並提供給主叫方。爾後,主叫方即可根據 UDP封包中被叫方I p位址,開始與被叫方互相傳送接收具560163 V. Description of the invention (3) For users using mobile communication devices, the communication cost of users cannot be reduced. [Objective and Summary of the Invention] In view of this, an object of the present invention is to provide a method for making an Internet call using a GPRS network. The method of the present invention is for a user of a mobile communication device to easily make an Internet call, and can reduce the call cost. According to the purpose of the present invention, a method for making an Internet call is provided. The method is used to provide a calling party to make an Internet call to a called party through a GPRS network, wherein the calling party and the called party The calling party has a calling party IP address and a called party IP address, respectively. This method starts with the calling party sending a call request message to the called party; wherein the request call message includes a caller IP address and a caller communication port (port). Then, the called party receives the call request message and learns the calling party's IP address and the calling party's communication port. Thereafter, the called party transcodes the user voice into a first voice data stream, and compresses the first voice data stream into at least one user data packet agreement (user da t agr am pr ot oco 1 UDP) packet, where the UDP packet has the IP address of the called party. Then, the called party sends this UDP packet to the calling party via the network according to the calling party's IP address and the calling party's communication port. The calling party receives the UDP packet through its calling party communication port, and performs unpacking and decompressing operations to form a first voice data stream. This second voice data stream is then decoded into the user voice and provided to the caller. After that, the calling party can start transmitting and receiving receivers to and from the called party according to the IP address of the called party in the UDP packet.
TW0559F(英華達).ptd 第6頁 560163 五、發明說明(4) 有語音數據之UDP封包以進行網路電話交談。 為讓本發明之上述目的、特徵、和優點能更明顯易 懂,下文特舉一較佳實施例,並配合所附圖式,作詳細說 明如下。 【較佳實施例】 目前多數之地區之行動電話系統皆採用北歐郵政及電 信組織(Nordic Post,Telephone and Telegraph)所提 出的一全球行動通訊系統(global system for mobile communication,GSM )。請參照第2圖,圖中GSM 201具有 基地台(base transceiver station,BTS ) 203 以及基地 台控制器(base station controller,BSC) 205 °BTS 203係用以負責GSM 201與行動通訊裝置間的無線訊號傳送 與接受,而BSC 205係用以管理與控制BTS 203。是故行動 通訊裴置係可發送無線訊號至BTS 203,BTS 203再透過 BSC 20 5將無線訊號傳送給GSM 201。如此,行動通訊裝置 即可利用GSM 2 0 1建立彼此間之無線連線以進行對話。 而如第2圖中所示,於GSM 20 1之網路架構下,歐洲電 信標準協會(European Telecommunication Standards Institute)研發整合封包無線電服務(general packet radio service,GPRS)網路211以提供數據傳輸的服務。 GPRS網路21 1之特色係為可將資料分封成數個封包 (packet ) —起傳送,因此其傳輸速度相當快速。 而本發明之一較佳實施例的撥打網路電話(InternetTW0559F (Yinghuada) .ptd Page 6 560163 V. Description of the invention (4) UDP packet with voice data for internet telephone conversation. In order to make the above-mentioned objects, features, and advantages of the present invention more comprehensible, a preferred embodiment is exemplified below and described in detail with reference to the accompanying drawings. [Preferred Embodiment] At present, mobile telephone systems in most regions use a global system for mobile communication (GSM) provided by the Nordic Post and Telephone and Telegraph. Please refer to Figure 2. In the figure, GSM 201 has a base transceiver station (BTS) 203 and a base station controller (BSC) 205 ° BTS 203 is responsible for wireless communication between GSM 201 and mobile communication devices. Signals are transmitted and received, and BSC 205 is used to manage and control BTS 203. This is why the mobile communication Pei Zhi can send wireless signals to BTS 203, and BTS 203 then sends the wireless signals to GSM 201 through BSC 20 5. In this way, the mobile communication devices can establish a wireless connection with each other for a conversation by using GSM 201. As shown in Figure 2, under the network architecture of GSM 201, the European Telecommunication Standards Institute has developed an integrated packet radio service (GPRS) network 211 to provide data transmission. service. The characteristic of GPRS network 21 1 is that data can be divided into several packets and transmitted together, so its transmission speed is quite fast. And a preferred embodiment of the present invention
TW0559F(英華達).ptd 第7頁 560163 五、發明說明(5) protocol phone)之方法係運用於第2圖之系統架構下, 其用以提供主叫方207藉由GPRS網路211撥打網路電話與被 叫方209進行對話。主叫方207與被叫方209係位於GSM 20 1 中,分別具有一電話號碼。因此,主叫方2 〇 7與被叫方2 〇 g 可利用GSM 20 1撥打對方之電話號碼以互相對話或傳送訊 息(譬如是短訊)。其中,主叫方2 〇 7與被叫方2 〇 9係可為 一行動通訊裝置’譬如是行動電話或具通訊功能之個人數 位助理(personal digital assistant,PDA)。 接著請同時參考第3圖,其所繪示為依照本發明一較 佳實施例之撥打網路電話之方法之流程圖。其中,步驟 301、步驟303、步驟305、步驟307與步驟309係為主叫方 207於本發明之方法中之作業流程。而步驟321、步驟步驟 323、步驟325、步驟327、步驟329與步驟331係為被叫方 2 0 9於本發明之方、法中之作業流程。 如第3圖所示,主叫方207與被叫方209分別於步驟3〇1 ' 與步驟321中開機登入GSM 201中。其中,主叫方2〇7更登 入GPRS網路211中,並獲得一主叫方網際網路(internet protocol, IP)位址。接著進行步驟303。 j 於步驟303中,主叫方207利用GSM 201發送一要求通 I話訊息至被叫方209之電話號碼。其中,此要求通話訊息 係為主叫方20 7要求被叫方20 9建立網路電話之連線要求 (request ),其包括一主叫方IP位址以及一主叫方通訊 端口( port )。接著,主叫方207進入步驟30 5中,傾聽 (listen)上述之主叫方通訊端口,以待接收來自被叫方TW0559F (Yinghuada) .ptd Page 7 560163 Fifth, the method of invention (5) protocol phone) method is applied to the system architecture in Figure 2, which is used to provide the calling party 207 dialing the network through GPRS network 211 The road phone has a conversation with the called party 209. The calling party 207 and the called party 209 are located in GSM 20 1 and each has a telephone number. Therefore, the calling party 207 and the called party 20 g can use GSM 20 1 to dial each other's phone number to talk to each other or send messages (such as SMS). Among them, the calling party 207 and the called party 109 may be a mobile communication device ', such as a mobile phone or a personal digital assistant (PDA) with a communication function. Please refer to FIG. 3 at the same time, which shows a flowchart of a method for making an Internet call according to a preferred embodiment of the present invention. Among them, step 301, step 303, step 305, step 307, and step 309 are the operation procedures of the calling party 207 in the method of the present invention. Step 321, step 323, step 325, step 327, step 329, and step 331 are the operations of the called party 209 in the method and method of the present invention. As shown in FIG. 3, the calling party 207 and the called party 209 are turned on and logged in to the GSM 201 in steps 301 ′ and 321, respectively. Among them, the calling party 207 also logs into the GPRS network 211 and obtains a calling party's Internet protocol (IP) address. Then, step 303 is performed. j In step 303, the calling party 207 uses GSM 201 to send a call request message to the telephone number of the called party 209. Among them, the call request message is a calling party 20 7 requesting the called party 20 9 to establish a connection request for an Internet phone (request), which includes a calling party IP address and a calling party communication port (port). . Next, the calling party 207 enters step 305, and listens to the communication port of the calling party, and waits to receive the call from the called party.
TW0559F(英華達).ptd 第8頁 560163TW0559F (Innova) .ptd Page 8 560163
五、發明說明(6) 209之回應。 而於步驟323中,被叫方209接收到主叫方207傳1^ 要求通話訊息,並獲知主叫方1 p位址及主叫方通訊端口, 接著,於步驟325中,被叫方20 9提供使用者決定是否進打 網路電話之連線。如果是’本發明之方法則進行步驟 3 2 7 ;否則,結束此方法。 > 當被叫方209之使用者同意與主叫方207進行網路電話 連線後,於步驟327中,被叫方209登入GPRS網路211 ’並 獲得一被叫方IP位址。隨後進行步驟329。 於步驟329中,被叫方329將使用者語音轉碼成一语曰 數據流,並將此語音數據流壓縮打包(pack )成至少一個5. Description of Invention (6) 209 Response. In step 323, the called party 209 receives the 1 ^ request message from the calling party 207, and learns the caller's 1 p address and the calling party's communication port. Then, in step 325, the called party 20 9 Provide users with a connection to decide whether to make an Internet call. If it is the method of the present invention, step 3 2 7 is performed; otherwise, the method ends. > After the user of the called party 209 agrees to make an Internet telephone connection with the calling party 207, in step 327, the called party 209 logs in to the GPRS network 211 'and obtains a called party IP address. Step 329 then follows. In step 329, the called party 329 transcodes the user's voice into a single-word data stream, and compresses the voice data stream into at least one
回應使用者資料包協定(user datagram protocol,UDP )封包。其中,此回應UDP封包係包括上述之被叫方IP位 址。而被叫方209依據主叫方IP位址及主叫方通訊端口且 藉由GPRS網路211傳送此UDP封包至主叫方207。 此時,主叫方207進入步驟307,經由主叫方通訊端口 接收被叫方20 9所傳送之UDP封包,並解開(unpack )、解 壓縮此UDP封包成一語音數摔流,再解碼此語音數據流成 使用者語音以提供給主叫方2 0 7之使用者。其中,主叫方 207藉由此UDP封包得知被叫方209之被叫方IP位址。 接著,主叫方20 7與被叫方209分別進行步驟30 9與步 驟331。主叫方20 7與被叫方209開始互相將其使用者語音 轉換成具有語音數據之UDP封包,並傳送給對方;且接收 來自對方之具有語音數據之UDP封包並轉換成使用者語Respond to user datagram protocol (UDP) packets. The response UDP packet includes the IP address of the called party. The called party 209 transmits the UDP packet to the calling party 207 via the GPRS network 211 according to the calling party's IP address and the calling party's communication port. At this time, the calling party 207 enters step 307, receives the UDP packet transmitted by the called party 209 through the calling party's communication port, and unpacks, decompresses the UDP packet into a voice stream, and then decodes this. The voice data is streamed into the user's voice and provided to the user of the calling party 207. Among them, the calling party 207 learns the called party IP address of the called party 209 through this UDP packet. Next, the calling party 20 7 and the called party 209 perform steps 30 9 and 331, respectively. The calling party 207 and the called party 209 start to convert their user voice into a UDP packet with voice data and send it to the other party; and receive a UDP packet with voice data from the other party and convert it into a user language
T1V0559F(英華達).ptdT1V0559F (Innova) .ptd
560163 五、發明說明(7) "----- =談藉此,主叫方207與被叫方20 9即可開始進行網路電話 、 而於步驟309與步驟331中,主叫方207或被叫方2〇9傳 送具有語音數據之UDP封包之方法,請參照第4圖。於第4 圖之步驟401中,使用者開始講話。接著,於步驟4〇3中, j據使用者預先設定之一音質參數而將使用者語音轉碼成 一語音數據流。此音質參數係用以決定語音數據流之音 質;而此音質參數可譬如是一取樣頻率(samplin^ ra"te )參數或一取樣解析度(resolution)參數。 接著於步驟4 05中,將上述之語音數據流進行壓縮並 打包成至少一個UDP封包。隨即,於步驟407中依據對方之 IP位址傳送此UDP封包。 另於步驟309與步驟331中,主叫方207或被叫方2〇9接 收處理具有語音數據之UDP封包之方法,請參照第5圖。於 第5圖之步驟5 〇1中,首先接收對方所傳送的具有語音數據 之UDP封包。接著進行步驟5〇3。 於步驟503中,解開上述之UDP封包並解壓縮成一語音 數據流。接著於步驟5 0 5中,將此語音數據流解碼成使用 者語音,並於步驟50 7中,播放此使用者語音。 【發明效果】 本發明上述實施例所揭露之一種撥打網路電話之方 法,其優點如下: (1 )節省行動通訊裝置使用者之通話費用:本發明560163 V. Description of the invention (7) " ----- = To talk about this, the calling party 207 and the called party 20 9 can start an internet call, and in steps 309 and 331, the calling party For the method of transmitting UDP packets with voice data at 207 or called party 209, please refer to FIG. 4. In step 401 of FIG. 4, the user starts speaking. Next, in step 403, j transcodes the user's voice into a voice data stream according to a voice quality parameter set in advance by the user. The sound quality parameter is used to determine the sound quality of the voice data stream; and the sound quality parameter may be, for example, a sampling frequency (samplin ^ ra " te) parameter or a sampling resolution parameter. Then in step 4 05, the above voice data stream is compressed and packed into at least one UDP packet. Then, in step 407, the UDP packet is transmitted according to the IP address of the other party. In step 309 and step 331, the calling party 207 or the called party 209 receives and processes the UDP packet with voice data. Please refer to FIG. 5. In step 5 of FIG. 5, a UDP packet with voice data transmitted by the other party is received first. Then proceed to step 503. In step 503, the above UDP packet is decompressed and decompressed into a voice data stream. Then, in step 505, the voice data stream is decoded into user voice, and in step 507, the user voice is played. [Effects of the Invention] The method for making a network call disclosed in the above embodiments of the present invention has the following advantages: (1) Saving the call cost of the user of the mobile communication device: the present invention
TTV0559F(英華達).ptd 第10頁 560163 五、發明說明(8) 利用GPRS網路來傳輸,而GPRS業者係依傳輸之數據量計 費,如此收費可實際反應使用者使用量,進而節省使用者 之花費。 (2 )簡化撥打網路電話之步驟:對使用者而言,僅 須發出一要求通話訊息,即可與對方建立網路電話之對 話,非常的便利。 (3 )自由調節通話品質:使用者可依其需要及GPRS 網路狀況,而自行設定音質參數,調整適合之通話品質。 (4 )連線容易:本發明之方法毋須撥打附加號碼, 亦無須通過isp的伺服器,而以點對點(peer—t〇—peer) 連線方式來實現網路電話之技術。 綜上所述,雖然本發明已以一較佳實施例揭露如上, 然其並非用以限定本發明,任何熟習此技藝者,在不脫離 本發明之精神和範圍内,當可作各種之更動與潤飾,因此 本發明之保冑m圍當視後附之申請專利冑圍所界定者為TTV0559F (Yinghuada) .ptd Page 10 560163 V. Description of the invention (8) Use GPRS network for transmission, and GPRS operators charge according to the amount of data transmitted. Such charges can actually reflect the user's usage and thus save usage. Costs. (2) Simplify the steps of making Internet calls: For users, they only need to send a call request message to establish an Internet call with the other party, which is very convenient. (3) Freely adjust call quality: Users can set their own sound quality parameters and adjust the appropriate call quality according to their needs and GPRS network conditions. (4) Easy connection: The method of the present invention does not need to dial an additional number, nor does it need to pass through an isp server, and implements the technology of Internet telephony in a peer-to-peer (peer-to-peer) connection. In summary, although the present invention has been disclosed as above with a preferred embodiment, it is not intended to limit the present invention. Any person skilled in the art can make various changes without departing from the spirit and scope of the present invention. And retouching, so the protection of the present invention is defined by the attached patent application as
560163560163
【圖式之簡單說明】 第1圖緣示為網路電話所運用之系統與網路的示意 圖。 第2圖緣不為行動通訊裝置與GSM &GPRS網路連接之網 路系統示意圖。 第3圖緣示為依照本發明一較佳實施例之撥打網路電 話之方法流程圖。 第4圖綠示為第3圖中之傳送具有語音數據之UDP封包 之步驟之方法流程圖。 第5圖繪示為第3圖中之接收具有語音數據之UDP封包 之步驟之方法流程圖。 【圖式標號說明】 100 :網際網路(Internet ) 1 〇 1, 1 0 5 :電腦 103, 107 :伺服器 111, 121 :電話 113,123 :公幕交換電話網路(public switched telephone network , PSTN ) 115, 125 :網際網路電話閘道器(Internet ! telephony gateway , ITG ) ! 201 :全球行動通訊系統(global system for mobile communication 5 GSM ) 203 :基地台(base transceiver station,BTS)[Brief description of the diagram] Figure 1 shows the schematic diagram of the system and network used by the Internet phone. Figure 2 is not a schematic diagram of the network system where the mobile communication device is connected to the GSM & GPRS network. Fig. 3 is a flow chart showing a method for making a network call according to a preferred embodiment of the present invention. Fig. 4 is a green flowchart of the method for transmitting the UDP packet with voice data in Fig. 3. Fig. 5 is a flowchart showing a method for receiving a UDP packet with voice data in Fig. 3. [Explanation of figure number] 100: Internet 1 〇1, 105: computer 103, 107: server 111, 121: telephone 113, 123: public switched telephone network (public switched telephone network, PSTN) 115, 125: Internet! Telephony gateway (ITG)! 201: Global system for mobile communication 5 GSM 203: Base transceiver station (BTS)
560163 圖式簡單說明 205 :基地台控制器(base station controller, BSC ) 2 0 7 :主叫方 2 0 9 :被叫方 211 :整合封包無線電服務(general packet radio service,GPRS)網路560163 Brief description of the diagram 205: Base station controller (BSC) 2 0 7: Calling party 2 0 9: Called party 211: Integrated packet radio service (GPRS) network
TW0559F(英華達).ptd 第13頁TW0559F (Innova) .ptd Page 13
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TW91106175A TW560163B (en) | 2002-03-28 | 2002-03-28 | Method for dialing Internet protocol phone |
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TW91106175A TW560163B (en) | 2002-03-28 | 2002-03-28 | Method for dialing Internet protocol phone |
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TW91106175A TW560163B (en) | 2002-03-28 | 2002-03-28 | Method for dialing Internet protocol phone |
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