TWM521851U - Internet telephone system - Google Patents

Internet telephone system Download PDF

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Publication number
TWM521851U
TWM521851U TW104219436U TW104219436U TWM521851U TW M521851 U TWM521851 U TW M521851U TW 104219436 U TW104219436 U TW 104219436U TW 104219436 U TW104219436 U TW 104219436U TW M521851 U TWM521851 U TW M521851U
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Taiwan
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call
network
voice
voice server
signal
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TW104219436U
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Chinese (zh)
Inventor
Wei Xu
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Weistech Technology Co Ltd
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Application filed by Weistech Technology Co Ltd filed Critical Weistech Technology Co Ltd
Priority to TW104219436U priority Critical patent/TWM521851U/en
Priority to CN201620341784.9U priority patent/CN205622716U/en
Publication of TWM521851U publication Critical patent/TWM521851U/en
Priority to JP2016002539U priority patent/JP3206362U/en

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Description

網路電話系統Internet telephony system

本創作係關於一種網路電話系統,特別是關於一種透過自動偵測網路訊號品質以決定將語音訊號轉入網際網路(Internet)或公眾交換電話網路(PSTN)之網路電話系統。This author is about an Internet telephony system, especially a network telephony system that automatically determines the quality of a network signal to transfer voice signals to the Internet or the Public Switched Telephone Network (PSTN).

一般傳統電話線路使用的網路是公眾交換電話網路(PSTN),傳送的是類比語音訊號,但,隨著網際網路蓬勃的發展,網路電話(VoIP)之使用日益普及,網路電話(VoIP)是透過IP網路傳輸數位語音資料技術的電話。網際網路協定(Internet Protocol,IP)原先是設計用來傳遞資料封包,而網路電話則是即時地在IP網路上傳遞數位語音訊號。網路電話(VoIP)語音傳送可以使用硬體或是軟體為架構,早期的產品是透過軟體架構之運作,使用者需要一台電腦執行軟體並與網際網路連線,電腦需要網路卡和麥克風。但連線常常只有半雙工,和傳統電話相較之下,談話過程像是在使用無線電通話,而非真正的電話,其通話品質不良,具改善空間。目前仍有許多網路電話(VoIP)產品使用軟體的架構,其優點是成本低廉。企業組織可能每個月都花費數千或上萬元不等的長途電話費,如果能透過固定成本低廉且可無限存取網際網路之網路電話,則通話費可大幅降低。The network used by traditional telephone lines is the Public Switched Telephone Network (PSTN), which transmits analog voice signals. However, with the rapid development of the Internet, the use of Internet Protocol (VoIP) is becoming more and more popular. (VoIP) is a telephone that transmits digital voice data technology over an IP network. The Internet Protocol (IP) was originally designed to deliver data packets, while Internet telephony is the immediate delivery of digital voice signals over IP networks. Voice over Internet Protocol (VoIP) voice transmission can be implemented in hardware or software. The early products were operated through a software architecture. Users need a computer to execute software and connect to the Internet. The computer needs a network card and microphone. However, the connection often only has half-duplex. Compared with the traditional telephone, the conversation process is like using a radio call instead of a real phone. The call quality is poor and the space is improved. There are still many software architectures for VoIP products that use software, which has the advantage of low cost. Organizational organizations may spend thousands or tens of thousands of long-distance telephone bills per month. If you can access Internet telephony with low fixed cost and unlimited access, the call charges can be greatly reduced.

雖然網路電話之使用費用低廉,但仍有其問題存在,網際網路協定(Internet Protocol,IP)是一種以封包(packet)為傳送基礎的協定,傳統電話使用之公眾交換電話網路(PSTN)則是採取線路交換(circuit switching)。以IP傳送語音資料時,系統會先將資料切割成封包在網路上傳送,由於IP的特性是盡力遞送(best-effort),並不提供保證服務,所以不同封包有可能採取不同路徑,造成封包順序錯亂或中途遺失,使得網路電話的語音品質無法獲得一定程度之保證。而使用公眾交換電話網路(PSTN)之線路交換技術則是在傳輸雙方之間建立固定路徑且保留必要頻寬的「線路」(circuit),因而語音品質較能夠得到保證。Although the use of Internet telephony is inexpensive, there are still problems. The Internet Protocol (IP) is a protocol based on packet transmission. The public switched telephone network (PSTN) used in traditional telephones. ) is to take circuit switching. When transmitting voice data by IP, the system will first cut the data into packets and transmit it on the network. Since the characteristics of IP are best-effort and do not provide guarantee service, different packets may take different paths, resulting in packets. The order is disordered or lost in the middle, so the voice quality of the Internet phone cannot be guaranteed to a certain extent. The circuit switching technology using the Public Switched Telephone Network (PSTN) is a "circuit" that establishes a fixed path between the transmitting parties and retains the necessary bandwidth, so that the voice quality can be guaranteed.

由以上之描述可知,網路電話(VoIP)與使用公眾交換電話網路(PSTN)之傳統電話各有其優缺點,如何保留兩者之優點並改善兩者之缺點應是該領域中急待解決之課題。As can be seen from the above description, both VoIP and traditional telephones using the Public Switched Telephone Network (PSTN) have their own advantages and disadvantages. How to preserve the advantages of both and improve the shortcomings of both should be urgent in the field. Solve the problem.

基於以上之陳述,本創作之主要目的即提供一種網路電話系統,該系統係包含:一手機、一手機撥號應用程式(APP)及一語音伺服器。該手機撥號應用程式(APP)安裝於該手機之上,該語音伺服器包括一RJ-45網路介面,一WIFI無線網路介面,該RJ-45網路介面及該WIFI無線網路介面用以連結區域網路(LAN)及網際網路(Internet);一RJ-11介面,一公眾交換電話網路(Public Switch Telephone Network/PSTN)撥號模組,一數位類比轉換模組,該RJ-11介面及該PSTN撥號模組用以連結PSTN網路,該數位類比轉換模組用以轉換數位類比訊號;一訊號偵測模組用以偵測網路訊號品質並決定將語音訊號轉入網際網路(Internet)或公眾交換電話網路(PSTN);一微處理器用以處理該語音伺服器之訊號;一通用輸入輸出(GPIO)介面用以連結其他裝置;一電源管理模組用以管理該語音伺服器之電源。當使用者使用本創作之網路電話系統時,其可使用安裝於其手機之一應用程式(APP)進行撥號,並透過手機上之WIFI模組透過區域網路(LAN)連結本創作網路電話系統之語音伺服器,該語音伺服器之訊號偵測模組會自動偵測網路訊號通話品質之好壞,若網際網路(Internet)通話品質好,則將語音訊號轉入網際網路(Internet),若公眾交換電話網路(PSTN)通話品質好,則將語音訊號轉入公眾交換電話網路(PSTN)。本創作所提供之網路電話系統具備相當之創新與進步,不但能透過固定成本低廉且可無限存取之網際網路進行語音通話,大幅降低通話費用,更可使用公眾交換電話網路(PSTN)之線路交換技術使得語音通話品質能夠得到保證,兩者兼顧,截長補短。使用類比線路(PSTN)通話品質更好更穩定,不受網路斷線影響,不受防火牆管制,不受頻寬限制。另外,使用本創作之網路電話系統,市話轉接至手機可透過WIFI,也可透過3G、4G、LTE、WIMAX使用區域更廣,人在國外,家中的電話也不會漏接,並可設定會議模式通話,一對多通話或三方通話,方便好用,市話透過網路通話可替代以前國際電話,通話費更省。本創作之網路電話系統亦可使用於平板電腦,平板電腦安裝APP就可接收轉接過來的市話。Based on the above statement, the main purpose of the present invention is to provide a network telephone system comprising: a mobile phone, a mobile phone dialing application (APP) and a voice server. The mobile phone dialing application (APP) is installed on the mobile phone. The voice server includes an RJ-45 network interface, a WIFI wireless network interface, the RJ-45 network interface and the WIFI wireless network interface. Connected to the local area network (LAN) and the Internet (Internet); an RJ-11 interface, a public switched telephone network (Public Switch Telephone Network/PSTN) dialing module, a digital analog conversion module, the RJ- The interface and the PSTN dialing module are used to connect to the PSTN network. The digital analog conversion module is used to convert the digital analog signal. The signal detection module is used to detect the quality of the network signal and decide to transfer the voice signal to the Internet. Internet (Internet) or Public Switched Telephone Network (PSTN); a microprocessor for processing the voice server signal; a general-purpose input and output (GPIO) interface for connecting other devices; and a power management module for managing The power of the voice server. When the user uses the created VoIP system, he can use one of the mobile phone applications (APP) to dial and connect to the authoring network via the local area network (LAN) via the WIFI module on the mobile phone. The voice server of the telephone system, the signal detection module of the voice server automatically detects the quality of the network signal call, and if the voice quality of the Internet (Internet) is good, the voice signal is transferred to the Internet. (Internet), if the public switched telephone network (PSTN) call quality is good, the voice signal is transferred to the public switched telephone network (PSTN). The VoIP system provided by this creation is quite innovative and progressive. It can not only make voice calls over the fixed-cost and unlimited access network, but also greatly reduce the cost of calls. It can also use the public switched telephone network (PSTN). The circuit switching technology enables the quality of voice calls to be guaranteed, both of which take care of each other. The use of analog line (PSTN) call quality is better and more stable, is not affected by network disconnection, is not controlled by the firewall, and is not limited by bandwidth. In addition, using the network phone system of this creation, the local call can be transferred to the mobile phone through WIFI, or the use area of 3G, 4G, LTE, WIMAX is wider, people are abroad, and the phone at home will not miss. It can also set conference mode call, one-to-many call or three-way call, which is convenient and easy to use. The local call can replace the previous international call through the network call, and the call charge is more economical. The VoIP phone system of this creation can also be used for tablets, and the tablet installation APP can receive the transferred local calls.

請參閱圖1,圖1係本創作之網路電話系統之系統架構示意圖。由圖1可知,本創作之網路電話系統01包括:一手機11、一手機撥號應用程式(APP)12及一語音伺服器13。該手機撥號應用程式(APP)12係安裝於該手機11之上,該語音伺服器13包括一RJ-45網路介面131,一WIFI無線網路介面132,該RJ-45網路介面131及該WIFI無線網路介面132用以連結區域網路(LAN)18及網際網路(Internet)14;一RJ-11介面133、一公眾交換電話網路(Public Switch Telephone Network/PSTN)撥號模組134,一數位類比轉換模組135,該RJ-11介面133及該PSTN撥號模組134用以連結PSTN網路15,該數位類比轉換模組135用以轉換數位類比訊號;一訊號偵測模組136用以偵測網路訊號之通話品質並決定將語音訊號轉入網際網路(Internet)14或公眾交換電話網路(PSTN)15;一微處理器137用以處理該語音伺服器13之各類訊號;一通用輸入輸出(GPIO)介面138用以連結其他裝置;一電源管理模組139用以管理該語音伺服器之電源。當使用者使用本創作之網路電話系統01時,其可使用安裝於其手機11之一應用程式(APP)12進行撥號,並透過手機上之WIFI模組透過區域網路(LAN)18連結本創作網路電話系統之語音伺服器13,該語音伺服器13之訊號偵測模組136會偵測網路訊號之通話品質之好壞,若網際網路(Internet)14通話品質好,則將語音訊號轉入網際網路(Internet)14,並與手機16連結通話,若公眾交換電話網路(PSTN)15通話品質好,則將語音訊號轉入公眾交換電話網路(PSTN)15,並與手機16連結通話。以上之實施例雖以手機11與手機16之連接通話為例,但不以此為限,手機11亦可應用上述之技術與流程與傳統PSTN電話17連接通話。Please refer to FIG. 1. FIG. 1 is a schematic diagram of the system architecture of the network telephone system of the present invention. As can be seen from FIG. 1, the network telephone system 01 of the present invention comprises: a mobile phone 11, a mobile phone dialing application (APP) 12 and a voice server 13. The mobile phone dialing application (APP) 12 is installed on the mobile phone 11. The voice server 13 includes an RJ-45 network interface 131, a WIFI wireless network interface 132, and the RJ-45 network interface 131 and The WIFI wireless network interface 132 is used to connect a local area network (LAN) 18 and an Internet (Internet) 14; an RJ-11 interface 133, a public switched telephone network (Public Switch Telephone Network/PSTN) dialing module 134, a digital analog conversion module 135, the RJ-11 interface 133 and the PSTN dialing module 134 are used to connect to the PSTN network 15, the digital analog conversion module 135 is used to convert the digital analog signal; The group 136 is configured to detect the call quality of the network signal and decide to transfer the voice signal to the Internet 14 or the Public Switched Telephone Network (PSTN) 15; a microprocessor 137 is configured to process the voice server 13 A variety of signals; a general purpose input and output (GPIO) interface 138 for connecting other devices; a power management module 139 for managing the power of the voice server. When the user uses the created VoIP system 01, it can be dialed using an application (APP) 12 installed on its mobile phone 11 and connected via a local area network (LAN) 18 via a WIFI module on the mobile phone. The voice server 13 of the authorizes the network telephone system, and the signal detecting module 136 of the voice server 13 detects the quality of the call of the network signal. If the voice quality of the Internet (Internet) 14 is good, The voice signal is transferred to the Internet 14 and connected to the mobile phone 16. If the public switched telephone network (PSTN) 15 has good call quality, the voice signal is transferred to the Public Switched Telephone Network (PSTN) 15. And connect to the phone 16 to talk. In the above embodiment, the connection between the mobile phone 11 and the mobile phone 16 is taken as an example, but not limited thereto, the mobile phone 11 can also use the above-mentioned technology and process to connect with the traditional PSTN telephone 17.

請參閱圖2,圖2係本創作之網路電話系統中手機應用程式(APP)與語音伺服器連接之示意圖。由圖2可知,當一使用者使用本系統通話時,本系統中手機應用程式(APP)與語音伺服器連接之流程步驟如下:Please refer to FIG. 2. FIG. 2 is a schematic diagram of a mobile application (APP) connected to a voice server in the network telephone system of the present invention. As can be seen from FIG. 2, when a user uses the system to make a call, the process steps of connecting the mobile application (APP) to the voice server in the system are as follows:

步驟21:使用者手機與語音伺服器建立連線;Step 21: The user's mobile phone establishes a connection with the voice server;

步驟22:使用者使用一手機應用程式(APP)撥打電話;Step 22: The user makes a call using a mobile application (APP);

步驟23:手機送出號碼至語音伺服器;Step 23: The mobile phone sends the number to the voice server;

步驟24:語音伺服器內一訊號偵測模組判斷網路通訊品質以決定語音訊號轉入網際網路或公眾交換電話網路;Step 24: A signal detection module in the voice server determines the quality of the network communication to determine whether the voice signal is transferred to the Internet or the public switched telephone network;

步驟25:確認與通話者接通;Step 25: Confirm that the caller is connected;

步驟26:送出與接收數位影音資料;Step 26: sending and receiving digital audio and video data;

步驟27:結束通話。Step 27: End the call.

請再參閱圖3,圖3係本創作之網路電話系統中手機接受來電通話與語音伺服器連接之示意圖。由圖3可知,當一使用者使用本系統接受來電通話時,本系統中手機與語音伺服器連接之流程步驟如下:Please refer to FIG. 3 again. FIG. 3 is a schematic diagram of the mobile phone receiving the incoming call and the voice server in the network telephone system of the present invention. It can be seen from FIG. 3 that when a user uses the system to accept an incoming call, the process steps of connecting the mobile phone to the voice server in the system are as follows:

步驟31:手機與語音伺服器建立連線;Step 31: The mobile phone establishes a connection with the voice server;

步驟32:接聽電話;Step 32: Answer the call;

步驟33:接受來自語音伺服器之通話;Step 33: accept the call from the voice server;

步驟34:送出與接收數位影音資料;Step 34: Sending and receiving digital audio and video data;

步驟35: 結束通話。Step 35: End the call.

請參閱圖4,圖4係本創作之網路電話系統中一般傳統電話撥至手機與語音伺服器連接之示意圖。由圖4可知,一般傳統電話撥至手機與語音伺服器連接流程步驟如下:Please refer to FIG. 4. FIG. 4 is a schematic diagram of a conventional telephone connection to a mobile phone and a voice server in the network telephone system of the present invention. As can be seen from Figure 4, the general procedure for connecting a traditional telephone to a mobile phone and a voice server is as follows:

步驟41:手機與語音伺服器建立連線;Step 41: The mobile phone establishes a connection with the voice server;

步驟42:接收雙音多頻(Dual-Tone Multi-Frequency/DTMF)語音撥號;Step 42: Receive Dual-Tone Multi-Frequency (DTMF) voice dialing;

步驟43:連接手機應用程式(APP);Step 43: Connect the mobile application (APP);

步驟44:建立通話;Step 44: Establish a call;

步驟45:接收與傳送來自一般傳統電話之類比語音訊號,接收與傳送來自手機之數位語音訊號;Step 45: receiving and transmitting an analog voice signal from a general conventional telephone, and receiving and transmitting a digital voice signal from the mobile phone;

步驟46:掛斷結束通話。Step 46: Hang up and end the call.

參閱圖5,圖5係本創作網路電話系統中手機撥至傳統電話與語音伺服器連接之示意圖。由圖5可知,手機撥至傳統電話與語音伺服器連接流程步驟如下:Referring to FIG. 5, FIG. 5 is a schematic diagram of a mobile phone dialing to a traditional telephone and a voice server in the author's network telephone system. As can be seen from Figure 5, the steps for connecting the mobile phone to the traditional telephone and voice server are as follows:

步驟51:手機與語音伺服器建立連線;Step 51: The mobile phone establishes a connection with the voice server;

步驟52:語音伺服器接收來自手機應用程式(APP)撥號;Step 52: The voice server receives the dialing from the mobile application (APP);

步驟53:依號碼產生雙音多頻(Dual-Tone Multi-Frequency/DTMF)語音撥號;Step 53: Generate Dual-Tone Multi-Frequency (DTMF) voice dialing according to the number;

步驟54:接通雙音多頻(Dual-Tone Multi-Frequency/DTMF)電話;Step 54: Turn on the Dual-Tone Multi-Frequency (DTMF) phone;

步驟55:建立通話;Step 55: Establish a call;

步驟56:接收與傳送來自手機之數位語音訊號,接收與傳送來自一般電話之類比語音訊號;Step 56: receiving and transmitting a digital voice signal from a mobile phone, and receiving and transmitting an analog voice signal from a general telephone;

步驟57:掛斷結束通話。Step 57: Hang up and end the call.

請參閱圖6,圖6係本創作網路電話系統中手機與語音伺服器連接之示意圖。由圖6可知,手機與語音伺服器連接流程步驟如下,其分為手機端及語音伺服器端:Please refer to FIG. 6. FIG. 6 is a schematic diagram of a connection between a mobile phone and a voice server in the author's network telephone system. It can be seen from FIG. 6 that the connection process between the mobile phone and the voice server is as follows: it is divided into a mobile phone terminal and a voice server terminal:

步驟611:手機端建立Socket;Step 611: The mobile terminal establishes a Socket;

步驟6111:語音伺服器端建立Socket;Step 6111: The voice server establishes a Socket;

步驟6112:語音伺服器端聆聽(Listen)是否有要求連接訊號進入;Step 6112: Whether the voice server listens (Listen) to request the connection signal to enter;

步驟6113:語音伺服器端接受連接訊號;Step 6113: The voice server end accepts the connection signal;

步驟614:手機連接語音伺服器;Step 614: The mobile phone is connected to the voice server.

步驟615:手機送出撥號與密碼,語音伺服器接收機號與密碼(步驟6151);Step 615: The mobile phone sends a dialing and password, a voice server receiver number and a password (step 6151);

步驟616:手機與語音伺服器建立連線(語音伺服器端與手機建立連線,步驟6161);Step 616: The mobile phone establishes a connection with the voice server (the voice server establishes a connection with the mobile phone, step 6161);

步驟617:手機等待與語音伺服器通話 (語音伺服器端等待與手機通話,步驟6171)。Step 617: The mobile phone waits for a call with the voice server (the voice server waits for a call with the mobile phone, step 6171).

由以上實施例之說明描述可知,本創作所提供之網路電話系統具備相當之創新與進步,不但能透過固定成本低廉且可無限存取之網際網路(Internet)進行語音通話,大幅降低通話費用,更可使用公眾交換電話網路(PSTN)之線路交換技術使得語音通話品質能夠得到保證,兩者兼顧,截長補短。使用類比線路(PSTN)通話品質更好更穩定,不受網路斷線影響,不受防火牆管制,不受頻寬限制。另外,使用本創作之網路電話系統,市話轉接至手機可透過WIFI,也可透過3G、4G、LTE、WIMAX使用區域更廣,人在國外,家中的電話也不會漏接,並可設定會議模式通話,一對多通話或三方通話,方便好用,市話透過網路通話可替代以前國際電話,通話費更省。本創作之網路電話系統亦可使用於平板電腦,平板電腦安裝APP就可接收轉接過來的市話。上列詳細說明係針對本創作之可行實施例之具體說明,惟該等實施例並非用以限制本創作之專利範圍,凡未脫離本創作技藝精神所為之等效實施或變更,均應包含於本案之專利範圍中。As can be seen from the description of the above embodiments, the VoIP system provided by the present invention has considerable innovation and progress, and can not only make voice calls through the fixed-cost and unlimited access Internet (Internet), but also greatly reduce the call. Costs, and the use of the public switched telephone network (PSTN) line switching technology can ensure the quality of voice calls, both of which take care of each other. The use of analog line (PSTN) call quality is better and more stable, is not affected by network disconnection, is not controlled by the firewall, and is not limited by bandwidth. In addition, using the network phone system of this creation, the local call can be transferred to the mobile phone through WIFI, or the use area of 3G, 4G, LTE, WIMAX is wider, people are abroad, and the phone at home will not miss. It can also set conference mode call, one-to-many call or three-way call, which is convenient and easy to use. The local call can replace the previous international call through the network call, and the call charge is more economical. The VoIP phone system of this creation can also be used for tablets, and the tablet installation APP can receive the transferred local calls. The detailed description above is for the specific description of the possible embodiments of the present invention, but the embodiments are not intended to limit the scope of the patents, and the equivalent implementations or modifications that are not included in the spirit of the present invention should be included in The patent scope of this case.

01‧‧‧網路電話系統
11‧‧‧手機
12‧‧‧手機撥號應用程式(APP)
13‧‧‧語音伺服器
131‧‧‧RJ-45網路介面
132‧‧‧WIFI無線網路介面
133‧‧‧RJ-11介面
134‧‧‧公眾交換電話網路(PSTN)撥號模組
135‧‧‧數位類比轉換模組
136‧‧‧訊號偵測模組
137‧‧‧微處理器
138‧‧‧通用輸入輸出(GPIO)介面
139‧‧‧電源管理模組
14‧‧‧網際網路(Internet)
15‧‧‧公眾交換電話網路(PSTN)網路
16‧‧‧手機
17‧‧‧PSTN電話
18‧‧‧區域網路(LAN)
21、22、23、24、25、26、27‧‧‧流程步驟
31、32、33、34、35‧‧‧流程步驟
41、42、43、44、45、46‧‧‧流程步驟
51、52、53、54、55、56、57‧‧‧流程步驟
611、614、615、616、616‧‧‧流程步驟
6111、6112、6113、6151、6161、6171‧‧‧流程步驟
01‧‧‧Internet phone system
11‧‧‧Mobile
12‧‧‧Mobile Dialing Application (APP)
13‧‧‧Voice Server
131‧‧‧RJ-45 network interface
132‧‧‧WIFI wireless network interface
133‧‧‧RJ-11 interface
134‧‧‧Public Switched Telephone Network (PSTN) Dialing Module
135‧‧‧Digital analog conversion module
136‧‧‧Signal Detection Module
137‧‧‧Microprocessor
138‧‧‧General Purpose Input Output (GPIO) Interface
139‧‧‧Power Management Module
14‧‧‧Internet (Internet)
15‧‧‧Public Exchange Telephone Network (PSTN) Network
16‧‧‧Mobile phones
17‧‧‧PSTN phone
18‧‧‧Local Network (LAN)
21, 22, 23, 24, 25, 26, 27 ‧ ‧ process steps
31, 32, 33, 34, 35‧‧‧ Process steps
41, 42, 43, 44, 45, 46‧‧‧ Process steps
51, 52, 53, 54, 55, 56, 57‧‧‧ process steps
611, 614, 615, 616, 616‧‧‧ process steps
6111, 6112, 6113, 6151, 6161, 6171‧‧‧ process steps

圖1            係本創作之網路電話系統之系統架構示意圖。 圖2            係本創作之網路電話系統中手機應用程式(APP)與語音伺服器連接之示意圖。 圖3            係本創作之網路電話系統中手機接受來電通話與語音伺服器連接之示意圖。 圖4            係本創作之網路電話系統中一般電話撥至手機與語音伺服器連接之示意圖。 圖5            係本創作之網路電話系統中手機撥至傳統電話與語音伺服器連接之示意圖。 圖6      係本創作之網路電話系統中手機與語音伺服器連接之示意圖。Figure 1 is a schematic diagram of the system architecture of the network telephone system of the present invention. Figure 2 is a schematic diagram of a mobile application (APP) connected to a voice server in the VoIP system of the present invention. Figure 3 is a schematic diagram of the mobile phone accepting an incoming call and a voice server connection in the network telephone system of the present invention. Figure 4 is a schematic diagram of a general telephone connection to a mobile phone and a voice server in the network telephone system of the present invention. Figure 5 is a schematic diagram of the connection of a mobile phone to a traditional telephone and a voice server in the VoIP system of the present invention. Figure 6 is a schematic diagram of the connection between the mobile phone and the voice server in the network telephone system of the present invention.

01‧‧‧網路電話系統 01‧‧‧Internet phone system

11‧‧‧手機 11‧‧‧Mobile

12‧‧‧手機撥號應用程式(APP) 12‧‧‧Mobile Dialing Application (APP)

13‧‧‧語音伺服器 13‧‧‧Voice Server

131‧‧‧RJ-45網路介面 131‧‧‧RJ-45 network interface

132‧‧‧WIFI無線網路介面 132‧‧‧WIFI wireless network interface

133‧‧‧RJ-11介面 133‧‧‧RJ-11 interface

134‧‧‧公眾交換電話網路(PSTN)撥號模組 134‧‧‧Public Switched Telephone Network (PSTN) Dialing Module

135‧‧‧數位類比轉換模組 135‧‧‧Digital analog conversion module

136‧‧‧訊號偵測模組 136‧‧‧Signal Detection Module

137‧‧‧微處理器 137‧‧‧Microprocessor

138‧‧‧通用輸入輸出(GPIO)介面 138‧‧‧General Purpose Input Output (GPIO) Interface

139‧‧‧電源管理模組 139‧‧‧Power Management Module

14‧‧‧網際網路(Internet) 14‧‧‧Internet (Internet)

15‧‧‧公眾交換電話網路(PSTN)網路 15‧‧‧Public Exchange Telephone Network (PSTN) Network

16‧‧‧手機 16‧‧‧Mobile phones

17‧‧‧PSTN電話 17‧‧‧PSTN phone

18‧‧‧區域網路(LAN) 18‧‧‧Local Network (LAN)

Claims (13)

一種網路電話系統,其包括:一通話裝置、一撥號應用程式(APP)及一語音伺服器,該撥號應用程式(APP)係安裝於該通話裝置之上,該語音伺服器包括: 一RJ-45網路介面、一WIFI無線網路介面,該RJ-45網路介面及該WIFI無線網路介面用以連結複數個網路; 一RJ-11介面、一公眾交換電話網路(Public Switch Telephone Network/PSTN)撥號模組、該RJ-11介面及該PSTN撥號模組用以連結PSTN網路; 一數位類比轉換模組用以轉換數位類比訊號; 一訊號偵測模組用以偵測網路訊號品質並決定將語音訊號轉入網際網路(Internet)或該公眾交換電話網路(PSTN);及 一微處理器用以處理該語音伺服器之各類訊號。A network telephone system includes: a call device, a dialing application (APP), and a voice server, the dialing application (APP) is installed on the call device, and the voice server includes: an RJ -45 network interface, a WIFI wireless network interface, the RJ-45 network interface and the WIFI wireless network interface for connecting a plurality of networks; an RJ-11 interface, a public switched telephone network (Public Switch) Telephone Network/PSTN) dialing module, the RJ-11 interface and the PSTN dialing module for connecting to the PSTN network; a digital analog conversion module for converting the digital analog signal; a signal detecting module for detecting The quality of the network signal determines the voice signal to be transferred to the Internet or the Public Switched Telephone Network (PSTN); and a microprocessor is used to process various signals of the voice server. 如請求項1所述之網路電話系統,其中,該語音伺服器可為一電腦。The network telephone system of claim 1, wherein the voice server is a computer. 如請求項1所述之網路電話系統,其中,該通話裝置可為一手機或一平板電腦。The network telephone system of claim 1, wherein the communication device is a mobile phone or a tablet computer. 如請求項1所述之網路電話系統,其中,該複數個網路可為網際網路(Internet)及區域網路(LAN)。The network telephone system of claim 1, wherein the plurality of networks are an Internet and a local area network (LAN). 如請求項1所述之網路電話系統,其中,該語音伺服器更可包括一通用輸入輸出(GPIO)介面用以連結其他裝置及一電源管理模組用以管理該語音伺服器之電源。The network telephone system of claim 1, wherein the voice server further comprises a universal input/output (GPIO) interface for connecting other devices and a power management module for managing power of the voice server. 如請求項1所述之網路電話系統,其中,一發話端轉接至該網路電話系統時,該發話端可透過以下之網路:PSTN、WIFI、3G、4G、LTE或WIMAX。The network telephone system of claim 1, wherein when the calling terminal is transferred to the network telephone system, the calling terminal can pass through the following networks: PSTN, WIFI, 3G, 4G, LTE or WIMAX. 如請求項1所述之網路電話系統,其中,使用該網路電話系統時可設定一會議通話模式,進行一對多通話或三方通話。The network telephone system according to claim 1, wherein when the VoIP system is used, a conference call mode can be set to perform one-to-many call or three-way call. 如請求項1所述之網路電話系統,當一使用者使用該網路電話系統時,其可使用該撥號應用程式(APP)進行撥號,並透過該通話裝置上之一WIFI模組透過一區域網路(LAN)連結該語音伺服器,該語音伺服器之訊號偵測模組會偵測網路訊號通話品質之好壞,若網際網路(Internet)通話品質好,則將語音訊號轉入該網際網路(Internet),若公眾交換電話網路(PSTN)通話品質好,則將語音訊號轉入公眾交換電話網路(PSTN)。The VoIP system of claim 1, wherein when a user uses the VoIP system, the dialing application (APP) can be used to dial and transmit through a WIFI module on the calling device. The local area network (LAN) connects to the voice server, and the voice signal detection module of the voice server detects the quality of the network signal call. If the voice quality of the Internet (Internet) is good, the voice signal is turned. Into the Internet (Internet), if the public switched telephone network (PSTN) call quality is good, the voice signal is transferred to the Public Switched Telephone Network (PSTN). 如請求項1所述之網路電話系統,其中,該撥號應用程式(APP)與該語音伺服器連接之流程步驟如下: 步驟1:通話裝置與語音伺服器建立連線; 步驟2:使用者使用該撥號應用程式(APP)撥打電話; 步驟3:通話裝置送出號碼至語音伺服器; 步驟4:語音伺服器內該訊號偵測模組判斷網路通訊品質以決定語音訊號轉入網際網路或公眾交換電話網路; 步驟5:確認與通話者接通; 步驟6:送出與接收數位影音資料; 步驟7: 結束通話。The network telephone system of claim 1, wherein the step of connecting the dialing application (APP) to the voice server is as follows: Step 1: The call device establishes a connection with the voice server; Step 2: The user Use the dialing application (APP) to make a call; Step 3: The call device sends the number to the voice server; Step 4: The signal detection module in the voice server determines the quality of the network communication to determine the voice signal to the Internet. Or the public exchange telephone network; Step 5: Confirm that the caller is connected; Step 6: Send and receive digital video data; Step 7: End the call. 如請求項1所述之網路電話系統,其中,當一使用者使用該系統接受來電通話時,該系統中通話裝置與語音伺服器連接之流程步驟如下: 步驟1: 通話裝置與語音伺服器建立連線; 步驟2:接聽電話; 步驟3:接受來自語音伺服器之通話; 步驟4:送出與接收數位影音資料; 步驟5: 結束通話。The network telephone system of claim 1, wherein when a user uses the system to accept an incoming call, the process of connecting the call device to the voice server in the system is as follows: Step 1: The call device and the voice server Establish a connection; Step 2: Answer the call; Step 3: Accept the call from the voice server; Step 4: Send and receive digital video and audio data; Step 5: End the call. 如請求項1所述之網路電話系統,其中,當一般傳統電話(PSTN)之通話裝置撥號至該網路電話系統之該通話裝置時,該語音伺服器連接流程步驟如下: 步驟1:通話裝置與語音伺服器建立連線; 步驟2:接收雙音多頻(Dual-Tone Multi-Frequency/DTMF)語音撥號; 步驟3:連接該通話裝置撥號應用程式(APP); 步驟4:建立通話; 步驟5:接收與傳送來自一般傳統電話之類比語音訊號,接收與傳送來自該通話裝置之數位語音訊號; 步驟6:掛斷結束通話。The network telephone system according to claim 1, wherein when the call device of the general conventional telephone (PSTN) dials to the communication device of the network telephone system, the voice server connection procedure is as follows: Step 1: Call The device establishes a connection with the voice server; Step 2: Receive Dual-Tone Multi-Frequency/DTMF voice dialing; Step 3: Connect the call device dialing application (APP); Step 4: Establish a call; Step 5: Receiving and transmitting an analog voice signal from a general conventional telephone, receiving and transmitting a digital voice signal from the communication device; Step 6: Hanging up the end call. 如請求項1所述之網路電話系統,其中,該通話裝置撥號至傳統電話(PSTN)與語音伺服器連接流程步驟如下: 步驟1:通話裝置與語音伺服器建立連線; 步驟2:語音伺服器接收來自該通話裝置撥號應用程式(APP)撥號; 步驟3:依號碼產生雙音多頻(Dual-Tone Multi-Frequency/DTMF)語音撥號; 步驟4:接通雙音多頻(Dual-Tone Multi-Frequency/DTMF)電話; 步驟5:建立通話; 步驟6:接收與傳送來自通話裝置之數位語音訊號,接收與傳送來自一般電話之類比語音訊號; 步驟7:掛斷結束通話。The network telephone system according to claim 1, wherein the step of connecting the call device to the traditional telephone (PSTN) and the voice server is as follows: Step 1: The call device establishes a connection with the voice server; Step 2: Voice The server receives the dialing application (APP) dialing from the calling device; Step 3: generates Dual-Tone Multi-Frequency/DTMF voice dialing according to the number; Step 4: Turn on the dual-tone multi-frequency (Dual- Tone Multi-Frequency/DTMF); Step 5: Establish a call; Step 6: Receive and transmit digital voice signals from the call device, receive and transmit analog voice signals from the general phone; Step 7: End the call. 如請求項1所述之網路電話系統,其中,該通話裝置與該語音伺服器連接流程步驟如下: 步驟1:該通話裝置端建立Socket,該語音伺服器端建立Socket; 步驟2:該語音伺服器端聆聽(Listen)是否有要求連接之訊號進入; 步驟3:該通話裝置發出連接訊號,該語音伺服器端接受連接訊號; 步驟4:該通話裝置連接該語音伺服器; 步驟5:該通話裝置送出撥號與密碼,該語音伺服器接收機號與密碼; 步驟6:該通話裝置與該語音伺服器建立連線; 步驟7:該通話裝置等待與該語音伺服器通話。The network telephone system according to claim 1, wherein the connection procedure between the communication device and the voice server is as follows: Step 1: The calling device establishes a Socket, and the voice server establishes a Socket; Step 2: the voice The server listens (Listen) whether there is a signal requesting connection; Step 3: The call device sends a connection signal, the voice server receives the connection signal; Step 4: The call device connects to the voice server; Step 5: The calling device sends the dialing and password, the voice server receiver number and the password; Step 6: the calling device establishes a connection with the voice server; Step 7: The calling device waits to talk to the voice server.
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