TWI448108B - System and method for video conference - Google Patents

System and method for video conference Download PDF

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TWI448108B
TWI448108B TW100109026A TW100109026A TWI448108B TW I448108 B TWI448108 B TW I448108B TW 100109026 A TW100109026 A TW 100109026A TW 100109026 A TW100109026 A TW 100109026A TW I448108 B TWI448108 B TW I448108B
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audio
buffer
tone multi
dual
signal
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TW201240388A (en
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chun yu Wang
Hsu Cheng Lin
Chien Hung Liu
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Ind Tech Res Inst
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視訊會議系統及方法Video conferencing system and method

本發明是有關於一種視訊會議系統及方法。The present invention relates to a video conferencing system and method.

電話從發明至今是非常重要的通訊工具。傳統電話藉由轉換收話端的電話號碼並發送一連串的雙音多頻(dual-tone multi-frequency,DTMF)訊號給公共交換電話網路(public switched telephone network,PSTN)的交換機,相關的交換機透過DTMF訊號了解收話端的位置並建立連線(call setup),之後兩端可以進行通話。其中連線建立的基礎在於DTMF訊號,而電信服務商附加的各種服務亦是在連線建立後,由使用者按下額外的數字按鍵,再由話機送出對應的DTMF訊號給服務端進行轉換而得以提供對應的服務。因此,DTMF訊號在整個電信網路中扮演了重要的角色。The phone has been a very important communication tool since its invention. The traditional telephone transmits the switch to the public switched telephone network (PSTN) by converting the telephone number of the receiving end and sending a series of dual-tone multi-frequency (DTMF) signals to the switch. The DTMF signal knows the location of the receiving end and establishes a call setup, after which the two ends can make a call. The basis for the establishment of the connection is the DTMF signal, and the various services attached by the telecommunication service provider are also after the connection is established, the user presses the extra numeric button, and then the corresponding DTMF signal is sent by the phone to the server for conversion. The corresponding service is available. Therefore, DTMF signals play an important role in the entire telecommunications network.

然而,隨著網際網路的蓬勃發展,網路電話(VoIP)逐漸取代傳統的通話方式。亦即,連線建立的基礎已由轉換收話端的電話號碼為DTMF訊號轉變為取得收話端的網際網路通訊協定(IP)位址並透過網際網路而建立連線。而網路電話或是通用序列匯排流(USB)電話轉接器所連接的傳統話機則演變成利用數字按鍵輸入IP垃址的工具,DTMF訊號則變成單純的數字訊號。However, with the rapid development of the Internet, VoIP has gradually replaced the traditional way of talking. That is to say, the basis of the connection establishment has been changed from the telephone number of the converted receiving terminal to the DTMF signal to obtain the Internet Protocol (IP) address of the receiving end and establish a connection through the Internet. The traditional telephone connected to the Internet telephony or the Universal Serial Bus (USB) telephone adapter has evolved into a tool for inputting IP addresses using digital buttons, and the DTMF signal becomes a simple digital signal.

本揭露是有關於一種視訊會議系統及方法,利用將雙音多頻訊號對應的字串命令轉換成鍵盤按鍵動作以控制應用程式,並偵測且消除音頻訊號中的雙音多頻訊號,故避免系統控制受到干擾而適用於視訊會議。The present disclosure relates to a video conferencing system and method for converting a string command corresponding to a dual tone multi-frequency signal into a keyboard button action to control an application, and detecting and eliminating a dual tone multi-frequency signal in the audio signal, Avoid system control interference and apply to video conferencing.

根據本揭露之第一方面,提出一種視訊會議系統,包括一話機、一電話轉接器以及一網路通訊裝置。話機用以產生一第一雙音多頻訊號。電話轉接器連接至話機,並用以偵測第一雙音多頻訊號且據以產生一第一命令字串。網路通訊裝置連接至電話轉接器與一網路,並用以依據一按鍵轉換表將第一命令字串轉換為一第一鍵盤按鍵動作以控制一應用程式。其中,網路通訊裝置更用以經由網路接收一第一音頻訊號,並偵測第一音頻訊號是否包括一第二雙音多頻訊號。若第一音頻訊號包括第二雙音多頻訊號,網路通訊裝置更用以依據偵測的結果將第一音頻訊號中對應第二雙音多頻訊號之部分進行靜音處理,並用以將靜音處理後之第一音頻訊號經由電話轉接器輸出至話機。According to a first aspect of the present disclosure, a video conferencing system is provided, including a telephone, a telephone adapter, and a network communication device. The phone is used to generate a first dual tone multi-frequency signal. The telephone adapter is connected to the phone, and is configured to detect the first dual tone multi-frequency signal and generate a first command string accordingly. The network communication device is connected to the telephone adapter and a network, and is configured to convert the first command string into a first keyboard button action according to a button conversion table to control an application. The network communication device is further configured to receive a first audio signal via the network, and detect whether the first audio signal includes a second dual tone multi-frequency signal. If the first audio signal includes the second dual tone multi-frequency signal, the network communication device is further configured to mute the portion of the first audio signal corresponding to the second dual tone multi-frequency signal according to the detected result, and to mute the signal. The processed first audio signal is output to the telephone via the telephone adapter.

根據本揭露之第二方面,提出一種視訊會議方法,應用於一視訊會議系統,視訊會議系統包括一話機、一電話轉接器及一網路通訊裝置。電話轉接器連接至話機,網路通訊裝置連接至電話轉接器與網路。視訊會議方法包括下列步驟。利用話機產生一第一雙音多頻訊號。利用電話轉接器偵測第一雙音多頻訊號且據以產生一第一命令字串。利用網路通訊裝置依據一按鍵轉換表將第一命令字串轉換為一第一鍵盤按鍵動作以控制一應用程式。利用網路通訊裝置經由網路接收一第一音頻訊號,並偵測第一音頻訊號是否包括一第二雙音多頻訊號。若第一音頻訊號包括第二雙音多頻訊號,利用網路通訊裝置依據偵測的結果將第一音頻訊號中對應第二雙音多頻訊號之部分進行靜音處理,並將靜音處理後之第一音頻訊號經由電話轉接器輸出至話機。According to a second aspect of the present disclosure, a video conferencing method is provided for use in a video conferencing system, the video conferencing system including a telephone, a telephone adapter, and a network communication device. The telephone adapter is connected to the telephone, and the network communication device is connected to the telephone adapter and the network. The video conferencing method includes the following steps. A first dual tone multi-frequency signal is generated by using the phone. The first dual tone multi-frequency signal is detected by the telephone adapter and a first command string is generated accordingly. The network communication device converts the first command string into a first keyboard button action according to a button conversion table to control an application. The network communication device receives a first audio signal via the network, and detects whether the first audio signal includes a second dual tone multi-frequency signal. If the first audio signal includes the second dual-tone multi-frequency signal, the network communication device mutes the portion of the first audio signal corresponding to the second dual-tone multi-frequency signal according to the detected result, and the mute processing is performed. The first audio signal is output to the phone via the telephone adapter.

為了對本揭露之上述及其他方面有更佳的瞭解,下文特舉一實施例,並配合所附圖式,作詳細說明如下:In order to better understand the above and other aspects of the present disclosure, an embodiment will be described hereinafter with reference to the accompanying drawings.

本揭露所提出之視訊會議系統及方法,利用將雙音多頻訊號對應的字串命令轉換成鍵盤按鍵動作以控制應用程式,並偵測且消除音頻訊號中的雙音多頻訊號,故避免系統控制受到干擾而適用於視訊會議。The video conferencing system and method proposed by the present disclosure converts a string command corresponding to a dual tone multi-frequency signal into a keyboard button action to control an application, and detects and eliminates a dual tone multi-frequency signal in the audio signal, thereby avoiding System control is interfered with for video conferencing.

請參照第1圖,其繪示依照一實施例之視訊會議系統之方塊圖。視訊會議系統100包括一話機110、一電話轉接器(phone adapter)120以及一網路通訊裝置(network communication device)130。電話轉接器120連接至話機110;網路通訊裝置130連接至電話轉接器120與一網路(未繪示於圖)。網路通訊裝置130例如為一主機(host)或一機上盒(set-top box),其例如利用一通用序列匯排流(USB)與電話轉接器120連接,然並不限制。網路通訊裝置130包括一網路模組140以及一音頻模組(audio module)150。網路模組140用以與網路建立通訊,故得以接收或傳送視訊會議相關的音頻訊號及視頻訊號。網路通訊裝置130實質上可接收來自話機110之音頻訊號或是從網路接收遠端視訊會議相關的音頻訊號。Please refer to FIG. 1 , which is a block diagram of a video conferencing system in accordance with an embodiment. The video conferencing system 100 includes a telephone 110, a telephone adapter 120, and a network communication device 130. The telephone adapter 120 is connected to the telephone 110; the network communication device 130 is connected to the telephone adapter 120 and a network (not shown). The network communication device 130 is, for example, a host or a set-top box, which is connected to the telephone adapter 120 by, for example, a universal serial bus (USB), but is not limited. The network communication device 130 includes a network module 140 and an audio module 150. The network module 140 is used to establish communication with the network, so that the audio signals and video signals related to the video conference can be received or transmitted. The network communication device 130 can substantially receive the audio signal from the phone 110 or receive the audio signal related to the far-end video conference from the network.

音頻模組150包括一雙音多頻(dual-tone multi-frequency,DTMF)訊號轉發(translation)模組162、一鍵盤事件發射器(keyboard event launcher)164、一雙音多頻消除(elimination)模組166、一音頻輸出模組168以及一音頻輸入模組169。音頻輸出模組168例如為一耳機,音頻輸入模組169為本機所接之實體裝置例如為一麥克風,音頻輸入模組169可接收來自網路的音頻封包而得到音頻訊號。於本實施例中,當視訊會議的連線建立後,話機110可作為控制器使用以透過電話轉接器120與網路通訊裝置130控制本地端的應用程式。請同時參照第2A圖及第2B圖,其分別繪示依照一實施例之視訊會議方法之部分流程圖。其中,分別繪示於第2A圖及第2B圖之視訊會議方法可同時進行,亦可以互為連續的行為,並不限制。於第2A圖之步驟S200中,按壓話機110的數字按鍵而產生一第一雙音多頻訊號。於步驟S210中,電話轉接器120偵測第一雙音多頻訊號且據以產生一第一命令字串。The audio module 150 includes a dual-tone multi-frequency (DTMF) signal forwarding module 162, a keyboard event launcher 164, and a dual-tone multi-frequency elimination (elimination). The module 166, an audio output module 168 and an audio input module 169. The audio output module 168 is, for example, a headphone. The audio input module 169 is a physical device connected to the device, for example, a microphone. The audio input module 169 can receive an audio packet from the network to obtain an audio signal. In this embodiment, after the connection of the video conference is established, the phone 110 can be used as a controller to control the local application through the phone adapter 120 and the network communication device 130. Please refer to FIG. 2A and FIG. 2B simultaneously, which respectively illustrate a partial flow chart of a video conference method according to an embodiment. The video conferencing methods respectively shown in FIG. 2A and FIG. 2B may be performed simultaneously, or may be continuous behaviors, and are not limited. In step S200 of FIG. 2A, the digital button of the phone 110 is pressed to generate a first dual tone multi-frequency signal. In step S210, the phone adapter 120 detects the first dual tone multi-frequency signal and generates a first command string accordingly.

於步驟S220中,利用雙音多頻訊號轉發模組162從電話轉接器接收120第一命令字串並將第一命令字串轉發至鍵盤事件發射器164。於步驟S230中,鍵盤事件發射器164依據一按鍵轉換表將第一命令字串轉換為第一鍵盤按鍵動作,本地端之一應用程式在收到第一鍵盤按鍵動作後即會執行對應的處理工作而改變使用者介面,故話機110可作為控制器控制本地端的應用程式。請參照第3圖,其繪示依照一實施例之按鍵轉換表之示意圖。於第3圖中,不同的話機數字按鍵對應至不同的鍵盤按鍵動作,例如話機數字按鍵「01」對應至鍵盤動作「Tab」,話機數字按鍵「08」對應至鍵盤動作「Down」等。此外亦可以將單一話機數字按鍵對應至組合鍵動作,但不限制,端視設計需求而定。In step S220, the dual command signal forwarding module 162 receives 120 the first command string from the phone adapter and forwards the first command string to the keyboard event transmitter 164. In step S230, the keyboard event transmitter 164 converts the first command string into the first keyboard button according to a button conversion table, and the application of the local end performs the corresponding processing after receiving the first keyboard button action. Work changes the user interface, so the phone 110 can be used as a controller to control the local application. Please refer to FIG. 3 , which illustrates a schematic diagram of a key conversion table according to an embodiment. In Figure 3, different phone numeric buttons correspond to different keyboard button actions. For example, the phone number button "01" corresponds to the keyboard action "Tab", and the phone number button "08" corresponds to the keyboard action "Down". In addition, a single phone number button can also be assigned to the combination key action, but it is not limited, depending on the design requirements.

當視訊會議進行中時,若遠端的使用者同時進行通話及利用話機控制操作系統,則本地端的視訊會議系統100在從網路接收視訊會議相關的視頻訊號與音頻訊號時,亦會聽到遠端的按鍵音輸出,亦即網路模組140從網路接收的音頻訊號會包括遠端送出的雙音多頻訊號,可能會導致電話轉接器120產生誤判而使得網路通訊裝置130產生錯誤控制本地端應用程式的情形發生。因此,於第2B圖之步驟S240中,網路通訊裝置130利用網路模組140經由網路接收一第一音頻訊號,此第一音頻訊號為來自遠端的視訊會議相關音頻訊號。第一音頻訊號在傳送前會被以脈碼調變(pulse code modulation,PCM)進行取樣,然後被壓縮切割為多個第一音頻封包以進行傳送,其中每秒脈碼取樣頻率例如為8000Hz。網路通訊裝置130在接收多個第一音頻封包後會進行解析及解壓縮而得到脈碼調變聲音片段,因此第一音頻訊號包括多個第一音頻片段。When the video conference is in progress, if the remote user simultaneously makes a call and uses the phone to control the operating system, the local videoconferencing system 100 also hears the video signal and the audio signal related to the video conference when receiving the video conference from the network. The button audio output of the terminal, that is, the audio signal received by the network module 140 from the network may include the dual-tone multi-frequency signal sent by the remote end, which may cause the telephone adapter 120 to generate a false positive, so that the network communication device 130 generates The error control of the local application occurs. Therefore, in step S240 of FIG. 2B, the network communication device 130 receives a first audio signal via the network by using the network module 140. The first audio signal is a video conference related audio signal from a remote end. The first audio signal is sampled by pulse code modulation (PCM) before being transmitted, and then compressed and cut into a plurality of first audio packets for transmission, wherein the pulse code sampling frequency per second is, for example, 8000 Hz. The network communication device 130 analyzes and decompresses the plurality of first audio packets to obtain a pulse code modulated sound segment, and thus the first audio signal includes a plurality of first audio segments.

之後,於步驟S250中,音頻輸入模組169偵測第一音頻訊號是否包括一第二雙音多頻訊號。音頻輸入模組169實際上會先偵測每個第一音頻片段之一DTMF旗標,若此旗標被設定為偽值(False),則此第一音頻片段不需被偵測而可直接被播放;若此旗標被設定為真值(True),則此第一音頻片段需被偵測後才能播放。音頻輸入模組169具有一緩衝器(buffer),其大小例如為10k~64k位元組(byte)。Then, in step S250, the audio input module 169 detects whether the first audio signal includes a second dual tone multi-frequency signal. The audio input module 169 actually detects one of the DTMF flags of each of the first audio segments. If the flag is set to a false value (False), the first audio segment does not need to be detected but can be directly Is played; if the flag is set to true (True), the first audio segment needs to be detected before it can be played. The audio input module 169 has a buffer whose size is, for example, 10k to 64k bytes.

音頻輸入模組169具有一第一緩衝器TMP_LINKED_LIST及一第二緩衝器Buffer。請參照第4圖,其繪示依照一實施例之音頻輸入模組之第一緩衝器及第二緩衝器之示意圖。於第4圖中,旗標被設定為真值的多個第一音頻片段(例如α03 )被依序複製並暫存在第一緩衝器TMP_LINKED_LIST。當第一緩衝器TMP_LINKED_LIST內之多個第一音頻片段之資料長度大於一最大偵測長度MaxDetectionLength,第一緩衝器TMP_LINKED_LIST內之多個第一音頻片段會被送到第二緩衝器Buffer內合併並得到對應之一位移值Offset,此位移值Offset為第一緩衝器TMP_LINKED_LIST內之多個第一音頻片段之資料長度和最大偵測長度MaxDetectionLength相除之餘數。The audio input module 169 has a first buffer TMP_LINKED_LIST and a second buffer Buffer. Please refer to FIG. 4 , which is a schematic diagram of a first buffer and a second buffer of an audio input module according to an embodiment. In FIG. 4, a plurality of first audio segments (for example, α 0 to α 3 ) whose flags are set to true values are sequentially copied and temporarily stored in the first buffer TMP_LINKED_LIST. When the data length of the plurality of first audio segments in the first buffer TMP_LINKED_LIST is greater than a maximum detection length MaxDetectionLength, the plurality of first audio segments in the first buffer TMP_LINKED_LIST are sent to the second buffer Buffer and merged. A corresponding displacement value Offset is obtained, and the displacement value Offset is a remainder divided by a data length of the plurality of first audio segments in the first buffer TMP_LINKED_LIST and a maximum detection length MaxDetectionLength.

音頻輸入模組169配合位移值Offset偵測第二緩衝器Buffer內之多個第一音頻片段是否包括第二雙音多頻訊號。亦即,音頻輸入模組169會偵測第二緩衝器Buffer內對應位置0到位置MaxDetectionLength的內容,並額外偵測第二緩衝器Buffer內對應位置Offset到位置(MaxDetectionLength+Offset)的內容。此外,音頻輸入模組169更反覆偵測第二緩衝器Buffer內之多個第一音頻片段是否包括第二雙音多頻訊號,且反覆偵測的次數K實質上可由話機110之一每秒脈碼調變取樣頻率(例如為8000Hz)、最大偵測長度MaxDetectionLength及一延遲範圍常數DelayConstant所決定,例如下述之公式(1),其中可藉由調整延遲範圍常數DelayConstant而得到最佳化之反覆偵測的次數K。The audio input module 169 cooperates with the displacement value Offset to detect whether the plurality of first audio segments in the second buffer Buffer include the second dual tone multi-frequency signal. That is, the audio input module 169 detects the content of the corresponding position 0 to the position MaxDetectionLength in the second buffer Buffer, and additionally detects the content of the corresponding position Offset to the position (MaxDetectionLength+Offset) in the second buffer Buffer. In addition, the audio input module 169 further detects whether the plurality of first audio segments in the second buffer Buffer include the second dual tone multi-frequency signal, and the number K of repeated detections is substantially one of the phones 110 per second. The pulse code modulation sampling frequency (for example, 8000 Hz), the maximum detection length MaxDetectionLength, and a delay range constant DelayConstant, for example, the following formula (1), which can be optimized by adjusting the delay range constant DelayConstant The number of times of repeated detection K.

K=ceil(8000/MaxDetectionLength)/DelayConstant (1)K=ceil(8000/MaxDetectionLength)/DelayConstant (1)

若音頻輸入模組169偵測到第二緩衝器Buffer內之多個第一音頻片段包括第二雙音多頻訊號,則於步驟S260中,雙音多頻消除模組166將第一音頻訊號中對應第二雙音多頻訊號之部分進行靜音處理。雙音多頻消除模阻166實質上將第二緩衝器Buffer內之多個第一音頻片段設定為0值,亦即第一音頻片段α03 均被填入0值。其中,被設定為0值之多個第一音頻片段的時間長度SkipCount實質上可由一延遲基本單位DelayEquivalent、話機110之一每秒脈碼調變取樣頻率(例如為8000Hz)及最大偵測長度MaxDetectionLength所決定,例如下述之公式(2)。If the audio input module 169 detects that the plurality of first audio segments in the second buffer buffer include the second dual tone multi-frequency signal, then in step S260, the dual tone multi-frequency cancellation module 166 sets the first audio signal. The part corresponding to the second dual tone multi-frequency signal is muted. The dual tone multi-frequency cancellation modulo 166 substantially sets a plurality of first audio segments in the second buffer Buffer to a value of 0, that is, the first audio segments α 0 to α 3 are all filled with a value of zero. The time length SkipCount of the plurality of first audio segments set to a value of 0 may be substantially a delay basic unit DelayEquivalent, one pulse per second of the phone 110, and the maximum detection length MaxDetectionLength. Determined, for example, the following formula (2).

SkipCount=ceil(8000/MaxDetectionLength)×DelayEquivalent (2)。SkipCount=ceil(8000/MaxDetectionLength)×DelayEquivalent (2).

接續於步驟S260之後,於步驟S270中,被雙音多頻消除模組166靜音處理後之第一音頻訊號輸出至音頻輸出模組168,或者被雙音多頻消除模組166靜音處理後之第一音頻訊號經由電話轉接器120輸出至話機110播放。而在步驟S250中,若音頻輸入模組169未偵測到第二緩衝器Buffer內之多個第一音頻片段包括第二雙音多頻訊號,則於步驟S280中,此些第一音頻片段會被直接播放。After the step S260, the first audio signal that has been muted by the dual-tone multi-frequency cancellation module 166 is output to the audio output module 168, or is muted by the dual-tone multi-frequency cancellation module 166. The first audio signal is output to the phone 110 via the phone adapter 120 for playback. In step S250, if the audio input module 169 does not detect that the plurality of first audio segments in the second buffer buffer include the second dual tone multi-frequency signal, then in step S280, the first audio segments are Will be played directly.

上述之步驟S240~S280係用以當視訊會議進行中時供本地端的視訊會議系統100濾除從遠端接收的視訊會議相關的音頻訊號中的按鍵音輸出(亦即雙音多頻訊號),以避免網路通訊裝置130產生錯誤控制本地端應用程式的情形發生。更進一步地,在本地端的視訊會議系統100輸出視訊會議相關的音頻訊號時,亦可先濾除音頻訊號中的按鍵音輸出以避免遠端產生誤動作。The above steps S240-S280 are used for the videoconferencing system 100 of the local end to filter out the button sound output (ie, the dual-tone multi-frequency signal) in the audio signal related to the video conference received from the remote end when the video conference is in progress. This prevents the network communication device 130 from generating an error to control the local end application. Further, when the videoconferencing system 100 of the local end outputs the audio signal related to the video conference, the button sound output in the audio signal may be filtered out first to prevent the remote terminal from malfunctioning.

於步驟S300中,音頻輸入模組169或話機110產生一第二音頻訊號。於步驟S310中,音頻輸入模組169偵測第二音頻訊號是否包括一第三雙音多頻訊號。若第二音頻訊號包括第三雙音多頻訊號,則於步驟S320中,雙音多頻消除模組166依據電話轉接器120偵測的結果將第二音頻訊號中對應第三雙音多頻訊號之部分進行靜音處理,並將靜音處理後之第二音頻訊號經由網路模組140輸出至網路。其中,音頻輸入模組169偵測雙音多頻訊號與雙音多頻消除模組166消除雙音多頻訊號的原理已詳述於步驟S250及S260的相關內容中,故於此不再重述。若第二音頻訊號未包括第三雙音多頻訊號,則於步驟S330中,直接將第二音頻訊號經由網路模組140輸出至網路。如此一來,本地端的視訊會議系統100即可輸出不包括按鍵音輸出(亦即雙音多頻訊號)的視訊會議相關的音頻訊號時,遠端的收話端即不會產生誤判而錯誤控制應用程式的情形。In step S300, the audio input module 169 or the phone 110 generates a second audio signal. In step S310, the audio input module 169 detects whether the second audio signal includes a third dual tone multi-frequency signal. If the second audio signal includes the third dual tone multi-frequency signal, then in step S320, the dual tone multi-frequency cancellation module 166 converts the third audio to the third audio signal according to the result detected by the phone adapter 120. The part of the frequency signal is muted, and the second audio signal after the mute processing is output to the network via the network module 140. The principle that the audio input module 169 detects the dual-tone multi-frequency signal and the dual-tone multi-frequency cancellation module 166 to eliminate the dual-tone multi-frequency signal is detailed in the related contents of steps S250 and S260, so it is not heavy here. Said. If the second audio signal does not include the third dual tone multi-frequency signal, then in step S330, the second audio signal is directly output to the network via the network module 140. In this way, when the local videoconferencing system 100 can output the audio signal related to the video conference that does not include the button sound output (that is, the dual tone multi-frequency signal), the remote receiving terminal does not generate false positives and is erroneously controlled. The case of the application.

本揭露上述實施例所揭露之視訊會議系統及方法,依據電話轉接器接收的雙音多頻訊號產生對應的字串命令,再將字串命令依據一按鍵轉換表轉換成鍵盤按鍵動作以控制應用程式,利於進行視訊會議。更進一步地,本揭露之視訊會議系統及方法偵測且消除從網路接收的音頻訊號及欲傳送至網路的音頻訊號中的雙音多頻訊號,故可避免本地端及遠端的視訊會議系統避免因誤判雙音多頻訊號而產生錯誤控制應用程式的情形。The video conference system and method disclosed in the above embodiments generate a corresponding string command according to the dual tone multi-frequency signal received by the telephone adapter, and then convert the string command into a keyboard button action according to a button conversion table to control The application facilitates video conferencing. Further, the video conferencing system and method of the present disclosure detects and eliminates the audio signals received from the network and the dual-tone multi-frequency signals in the audio signals to be transmitted to the network, thereby avoiding local and remote video recording. The conference system avoids situations in which an error control application is generated due to misjudgment of the DTMF signal.

綜上所述,雖然本發明已以多個實施例揭露如上,然其並非用以限定本發明。本發明所屬技術領域中具有通常知識者,在不脫離本發明之精神和範圍內,當可作各種之更動與潤飾。因此,本發明之保護範圍當視後附之申請專利範圍所界定者為準。In the above, the present invention has been disclosed in the above embodiments, but it is not intended to limit the present invention. A person skilled in the art can make various changes and modifications without departing from the spirit and scope of the invention. Therefore, the scope of the invention is defined by the scope of the appended claims.

100...視訊會議系統100. . . Video conferencing system

110...話機110. . . Telephone

120...電話轉接器120. . . Telephone adapter

130...網路通訊裝置130. . . Network communication device

140...網路模組140. . . Network module

150...音頻模組150. . . Audio module

162...雙音多頻訊號轉發模組162. . . Dual tone multi-frequency signal forwarding module

164...鍵盤事件發射器164. . . Keyboard event emitter

166...雙音多頻消除模組166. . . Dual tone multi-frequency cancellation module

168...音頻輸出模組168. . . Audio output module

169...音頻輸入模組169. . . Audio input module

TMP_LINKED_LIST...第一緩衝器TMP_LINKED_LIST. . . First buffer

Buffer...第二緩衝器Buffer. . . Second buffer

第1圖繪示依照一實施例之視訊會議系統之方塊圖。FIG. 1 is a block diagram of a video conferencing system in accordance with an embodiment.

第2A圖及第2B圖分別繪示依照一實施例之視訊會議方法之部分流程圖。2A and 2B are respectively a partial flow chart of a video conference method according to an embodiment.

第3圖繪示依照一實施例之按鍵轉換表之示意圖。FIG. 3 is a schematic diagram of a key conversion table according to an embodiment.

第4圖繪示依照一實施例之音頻輸入模組之第一緩衝器及第二緩衝器之示意圖。FIG. 4 is a schematic diagram of a first buffer and a second buffer of an audio input module according to an embodiment.

100...視訊會議系統100. . . Video conferencing system

110...話機110. . . Telephone

120...電話轉接器120. . . Telephone adapter

130...網路通訊裝置130. . . Network communication device

140...網路模組140. . . Network module

150...音頻模組150. . . Audio module

162...雙音多頻訊號轉發模組162. . . Dual tone multi-frequency signal forwarding module

164...鍵盤事件發射器164. . . Keyboard event emitter

166...雙音多頻消除模組166. . . Dual tone multi-frequency cancellation module

168...音頻輸出模組168. . . Audio output module

169...音頻輸入模組169. . . Audio input module

Claims (18)

一種視訊會議系統,包括:一話機,用以產生一第一雙音多頻訊號;一電話轉接器,連接至該話機,並用以偵測該第一雙音多頻訊號且據以產生一第一命令字串;以及一網路通訊裝置,連接至該電話轉接器與一網路,並用以依據一按鍵轉換表將該第一命令字串轉換為一第一鍵盤按鍵動作以控制一應用程式;其中,該網路通訊裝置更用以經由該網路接收一第一音頻訊號,並偵測該第一音頻訊號是否包括一第二雙音多頻訊號,若該第一音頻訊號包括該第二雙音多頻訊號,該網路通訊裝置更用以依據偵測的結果將該第一音頻訊號中對應該第二雙音多頻訊號之部分進行靜音處理,並用以將靜音處理後之該第一音頻訊號經由該電話轉接器輸出至該話機。A video conferencing system includes: a phone for generating a first dual tone multi-frequency signal; a telephone adapter connected to the phone for detecting the first dual tone multi-frequency signal and generating a a first command string; and a network communication device connected to the phone adapter and a network, and configured to convert the first command string into a first keyboard button action according to a button conversion table to control one An application, wherein the network communication device is further configured to receive a first audio signal via the network, and detect whether the first audio signal includes a second dual tone multi-frequency signal, if the first audio signal includes The second dual-tone multi-frequency signal is further configured to mute the portion of the first audio signal corresponding to the second dual-tone multi-frequency signal according to the detected result, and use the mute processing The first audio signal is output to the phone via the phone adapter. 如申請專利範圍第1項所述之視訊會議系統,其中該網路通訊裝置包括:一網路模組,用以與該網路建立通訊;以及一音頻模組,包括:一鍵盤事件發射器,用以依據該按鍵轉換表將該第一命令字串轉換為該第一鍵盤按鍵動作;一雙音多頻訊號轉發模組,用以從該電話轉接器接收該第一命令字串並將該第一命令字串轉發至該鍵盤事件發射器;一音頻輸入模組,用以經由該網路接收該第一音頻訊號,並偵測該第一音頻訊號是否包括該第二雙音多頻訊號;一雙音多頻消除模組,用以將該第一音頻訊號中對應該第二雙音多頻訊號之部分進行靜音處理;以及一音頻輸出模組,用以將靜音處理後之該第一音頻訊號經由該電話轉接器輸出至該話機。The video conferencing system of claim 1, wherein the network communication device comprises: a network module for establishing communication with the network; and an audio module comprising: a keyboard event transmitter The first command string is converted into the first keyboard button action according to the button conversion table; a dual tone multi-frequency signal forwarding module is configured to receive the first command string from the phone adapter and Forwarding the first command string to the keyboard event transmitter; an audio input module for receiving the first audio signal via the network, and detecting whether the first audio signal includes the second dual tone a frequency multi-frequency cancellation module for muting the portion of the first audio signal corresponding to the second dual-tone multi-frequency signal; and an audio output module for performing the mute processing The first audio signal is output to the phone via the phone adapter. 如申請專利範圍第2項所述之視訊會議系統,其中該第一音頻訊號包括複數個第一音頻片段,該音頻輸入模組具有一第一緩衝器及一第二緩衝器,該第一緩衝器用以依序複製並暫存該些第一音頻片段,當該第一緩衝器內之該些第一音頻片段之資料長度大於一最大偵測長度,該第一緩衝器內之該些第一音頻片段於該第二緩衝器內被合併並得到對應之一位移值,該音頻輸入模組配合該位移值偵測該第二緩衝器內之該些第一音頻片段是否包括該第二雙音多頻訊號,若該些第一音頻片段包括該第二雙音多頻訊號,該雙音多頻消除模組將該第二緩衝器內之該些第一音頻片段設定為0值。The video conferencing system of claim 2, wherein the first audio signal comprises a plurality of first audio segments, the audio input module has a first buffer and a second buffer, the first buffer The device is configured to sequentially copy and temporarily store the first audio segments. When the data length of the first audio segments in the first buffer is greater than a maximum detection length, the first ones in the first buffer And the audio input module is combined with the displacement value to detect whether the first audio segments in the second buffer include the second dual tone. The multi-tone signal, if the first audio segment includes the second dual-tone multi-frequency signal, the dual-tone multi-frequency cancellation module sets the first audio segments in the second buffer to a value of zero. 如申請專利範圍第3項所述之視訊會議系統,其中該音頻輸入模組更反覆偵測該第二緩衝器內之該些第一音頻片段是否包括該第二雙音多頻訊號,且反覆偵測的次數由該話機之一每秒脈碼調變取樣頻率、該最大偵測長度及一延遲範圍常數所決定。The video conferencing system of claim 3, wherein the audio input module further detects whether the first audio segments in the second buffer include the second dual tone multi-frequency signal, and repeatedly The number of detections is determined by one of the phone's pulse code modulation sampling frequency per second, the maximum detection length, and a delay range constant. 如申請專利範圍第3項所述之視訊會議系統,其中被設定為0值之該些第一音頻片段的時間長度由一延遲基本單位、該話機之一每秒脈碼調變取樣頻率及該最大偵測長度所決定。The video conferencing system of claim 3, wherein the length of time of the first audio segments set to a value of 0 is determined by a delay basic unit, one of the phones, and a pulse code modulation sampling frequency per second. The maximum detection length is determined. 如申請專利範圍第2項所述之視訊會議系統,其中該話機更用以產生一第二音頻訊號,該音頻輸入模組更用以偵測該第二音頻訊號是否包括一第三雙音多頻訊號,若該第二音頻訊號包括該第三雙音多頻訊號,該雙音多頻消除模組更用以依據該音頻輸入模組偵測的結果將該第二音頻訊號中對應該第三雙音多頻訊號之部分進行靜音處理,並用以將靜音處理後之該第二音頻訊號經由該網路模組輸出至該網路。The video conferencing system of claim 2, wherein the phone is further configured to generate a second audio signal, wherein the audio input module is further configured to detect whether the second audio signal includes a third dual tone a frequency signal, if the second audio signal includes the third dual tone multi-frequency signal, the dual tone multi-frequency cancellation module is further configured to correspond to the second audio signal according to the result detected by the audio input module The part of the three-tone multi-frequency signal is muted, and is used to output the second audio signal after the mute processing to the network via the network module. 如申請專利範圍第6項所述之視訊會議系統,其中該第二音頻訊號包括複數個第二音頻片段,該該音頻輸入模組具有一第一緩衝器及一第二緩衝器,該第一緩衝器用以依序複製並暫存該些第二音頻片段,當該第一緩衝器內之該些第二音頻片段之資料長度大於一最大偵測長度,該第一緩衝器內之該些第二音頻片段於該第二緩衝器內被合併並得到對應之一位移值,該音頻輸入模組配合該位移值偵測該第二緩衝器內之該些第二音頻片段是否包括該第三雙音多頻訊號,若該些第二音頻片段包括該第三雙音多頻訊號,該雙音多頻消除模組將該第二緩衝器內之該些第二音頻片段設定為0值。The video conferencing system of claim 6, wherein the second audio signal comprises a plurality of second audio segments, the audio input module having a first buffer and a second buffer, the first The buffer is configured to sequentially copy and temporarily store the second audio segments. When the data length of the second audio segments in the first buffer is greater than a maximum detection length, the first The second audio segment is combined in the second buffer to obtain a corresponding one of the displacement values, and the audio input module cooperates with the displacement value to detect whether the second audio segments in the second buffer include the third pair The multi-tone signal, if the second audio segment includes the third dual-tone multi-frequency signal, the dual-tone multi-frequency cancellation module sets the second audio segments in the second buffer to a value of zero. 如申請專利範圍第7項所述之視訊會議系統,其中該音頻輸入模組更反覆偵測該第二緩衝器內之該些第二音頻片段是否包括該第三雙音多頻訊號,且反覆偵測的次數由該話機之一每秒脈碼調變取樣頻率、該最大偵測長度及一延遲範圍常數所決定。The video conferencing system of claim 7, wherein the audio input module further detects whether the second audio segments in the second buffer include the third dual tone multi-frequency signal, and repeatedly The number of detections is determined by one of the phone's pulse code modulation sampling frequency per second, the maximum detection length, and a delay range constant. 如申請專利範圍第7項所述之視訊會議系統,其中被設定為0值之該些第二音頻片段的時間長度由一延遲基本單位、該話機之一每秒脈碼調變取樣頻率及該最大偵測長度所決定。The video conferencing system of claim 7, wherein the length of time of the second audio segments set to a value of 0 is determined by a delay basic unit, one of the phones, and a pulse code modulation sampling frequency per second. The maximum detection length is determined. 一種視訊會議方法,應用於一視訊會議系統,該視訊會議系統包括一話機、一電話轉接器及一網路通訊裝置,該電話轉接器連接至該話機,該網路通訊裝置連接至該電話轉接器與一網路,該視訊會議方法包括:利用該話機產生一第一雙音多頻訊號;利用該電話轉接器偵測該第一雙音多頻訊號且據以產生一第一命令字串;利用該網路通訊裝置依據一按鍵轉換表將該第一命令字串轉換為一第一鍵盤按鍵動作以控制一應用程式;利用該網路通訊裝置經由該網路接收一第一音頻訊號,並偵測該第一音頻訊號是否包括一第二雙音多頻訊號;若該第一音頻訊號包括該第二雙音多頻訊號,利用該網路通訊裝置依據偵測的結果將該第一音頻訊號中對應該第二雙音多頻訊號之部分進行靜音處理,並將靜音處理後之該第一音頻訊號經由該電話轉接器輸出至該話機。A video conferencing method is applied to a video conferencing system, the video conferencing system includes a telephone, a telephone adapter, and a network communication device, the telephone adapter is connected to the telephone, and the network communication device is connected to the a telephone adapter and a network, the video conference method includes: generating a first dual tone multi-frequency signal by using the phone; detecting the first dual tone multi-frequency signal by using the telephone adapter, and generating a first a command string; the network communication device converts the first command string into a first keyboard button action according to a button conversion table to control an application; and the network communication device receives a first via the network An audio signal, and detecting whether the first audio signal includes a second dual tone multi-frequency signal; if the first audio signal includes the second dual tone multi-frequency signal, using the network communication device according to the detected result The part of the first audio signal corresponding to the second dual tone multi-frequency signal is muted, and the first audio signal after the mute processing is output to the phone via the telephone adapter. 如申請專利範圍第10項所述之視訊會議方法,其中該網路通訊裝置包括一網路模組及一音頻模組,該網路模組用以與該網路建立通訊,該音頻模組包括一鍵盤事件發射器、一雙音多頻訊號轉發模組、一音頻輸入模組、一雙音多頻消除模組及一音頻輸出模組,該視訊會議方法更包括:利用該雙音多頻訊號轉發模組從該電話轉接器接收該第一命令字串並將該第一命令字串轉發至該鍵盤事件發射器;利用該鍵盤事件發射器依據該按鍵轉換表將該第一命令字串轉換為該第一鍵盤按鍵動作;利用該音頻輸入模組經由該網路接收該第一音頻訊號,並偵測該第一音頻訊號是否包括該第二雙音多頻訊號;利用該雙音多頻消除模組將該第一音頻訊號中對應該第二雙音多頻訊號之部分進行靜音處理;以及利用該音頻輸出模組將靜音處理後之該第一音頻訊號經由該電話轉接器輸出至該話機。The video conferencing method of claim 10, wherein the network communication device comprises a network module and an audio module, wherein the network module is configured to establish communication with the network, the audio module The device includes a keyboard event transmitter, a dual-tone multi-frequency signal forwarding module, an audio input module, a dual-tone multi-frequency cancellation module, and an audio output module. The video conference method further includes: utilizing the dual tone The frequency forwarding module receives the first command string from the phone adapter and forwards the first command string to the keyboard event transmitter; using the keyboard event transmitter to use the key conversion table to the first command Converting the string to the first keyboard button action; receiving, by the audio input module, the first audio signal via the network, and detecting whether the first audio signal includes the second dual tone multi-frequency signal; The audio multi-frequency cancellation module mutes the portion of the first audio signal corresponding to the second dual-tone multi-frequency signal; and uses the audio output module to rotate the first audio signal after the mute processing via the telephone The output to the telephone. 如申請專利範圍第11項所述之視訊會議方法,其中該第一音頻訊號包括複數個第一音頻片段,該音頻輸入模組具有一第一緩衝器及一第二緩衝器,該視訊會議方法更包括:利用第一該緩衝器依序複製並暫存該些第一音頻片段;當該第一緩衝器內之該些第一音頻片段之資料長度大於一最大偵測長度,利用該第二緩衝器合併該第一緩衝器內之該些第一音頻片段並得到對應之一位移值;利用該音頻輸入模組配合該位移值偵測該第二緩衝器內之該些第一音頻片段是否包括該第二雙音多頻訊號;以及若該些第一音頻片段包括該第二雙音多頻訊號,利用該雙音多頻消除模組將該第二緩衝器內之該些第一音頻片段設定為0值。The video conferencing method of claim 11, wherein the first audio signal comprises a plurality of first audio segments, the audio input module has a first buffer and a second buffer, and the video conference method The method further includes: sequentially copying and temporarily storing the first audio segments by using the first buffer; and using the second when the data length of the first audio segments in the first buffer is greater than a maximum detection length The buffer combines the first audio segments in the first buffer and obtain a corresponding one of the displacement values; and the audio input module cooperates with the displacement value to detect whether the first audio segments in the second buffer are The second dual tone multi-frequency signal is included; and if the first audio segment includes the second dual tone multi-frequency signal, the first audio in the second buffer is used by the dual tone multi-frequency cancellation module The clip is set to a value of 0. 如申請專利範圍第12項所述之視訊會議方法,更包括:利用該音頻輸入模組反覆偵測該第二緩衝器內之該些第一音頻片段是否包括該第二雙音多頻訊號,且反覆偵測的次數由該話機之一每秒脈碼調變取樣頻率、該最大偵測長度及一延遲範圍常數所決定。The video conferencing method of claim 12, further comprising: detecting, by the audio input module, whether the first audio segments in the second buffer comprise the second dual tone multi-frequency signal, The number of times of repeated detection is determined by one pulse code modulation sampling frequency per second of the phone, the maximum detection length and a delay range constant. 如申請專利範圍第12項所述之視訊會議方法,其中被設定為0值之該些第一音頻片段的時間長度由一延遲基本單位、該話機之一每秒脈碼調變取樣頻率及該最大偵測長度所決定。The video conferencing method of claim 12, wherein the length of time of the first audio segments set to a value of 0 is determined by a delay basic unit, one pulse per second of the telephone, and a sampling frequency of the pulse code. The maximum detection length is determined. 如申請專利範圍第11項所述之視訊會議方法,更包括:利用該音頻輸入模組或該話機產生一第二音頻訊號;利用該音頻輸入模組偵測該第二音頻訊號是否包括一第三雙音多頻訊號;以及若該第二音頻訊號包括該第三雙音多頻訊號,利用該雙音多頻消除模組依據該音頻輸入模組偵測的結果將該第二音頻訊號中對應該第三雙音多頻訊號之部分進行靜音處理,並用以將靜音處理後之該第二音頻訊號經由該網路模組輸出至該網路。The video conferencing method of claim 11, further comprising: generating a second audio signal by using the audio input module or the phone; and detecting, by the audio input module, whether the second audio signal includes a first a dual-tone multi-frequency signal; and if the second audio signal includes the third dual-tone multi-frequency signal, the dual-tone multi-frequency cancellation module is used to detect the second audio signal according to the result detected by the audio input module The part of the third dual-tone multi-frequency signal is muted, and the second audio signal after the mute processing is output to the network via the network module. 如申請專利範圍第15項所述之視訊會議方法,其中該第二音頻訊號包括複數個第二音頻片段,該音頻輸入模組具有一第一緩衝器及一第二緩衝器,該視訊會議方法更包括:利用該第一緩衝器依序複製並暫存該些第二音頻片段;當該第一緩衝器內之該些第二音頻片段之資料長度大於一最大偵測長度,利用該第二緩衝器合併該第一緩衝器內之該些第二音頻片段並得到對應之一位移值;利用該音頻輸入模組配合該位移值偵測該第二緩衝器內之該些第二音頻片段是否包括該第三雙音多頻訊號;以及若該些第二音頻片段包括該第三雙音多頻訊號,利用該雙音多頻消除模組將該第二緩衝器內之該些第二音頻片段設定為0值。The video conferencing method of claim 15, wherein the second audio signal comprises a plurality of second audio segments, the audio input module has a first buffer and a second buffer, and the video conference method The method further includes: sequentially copying and temporarily storing the second audio segments by using the first buffer; and using the second when the data length of the second audio segments in the first buffer is greater than a maximum detection length The buffer combines the second audio segments in the first buffer and obtain a corresponding one of the displacement values; and the audio input module cooperates with the displacement value to detect whether the second audio segments in the second buffer are The third dual tone multi-frequency signal is included; and if the second audio segment includes the third dual tone multi-frequency signal, the second audio in the second buffer is used by the dual tone multi-frequency cancellation module The clip is set to a value of 0. 如申請專利範圍第16項所述之視訊會議方法,其中該音頻輸入模組更反覆偵測該第二緩衝器內之該些第二音頻片段是否包括該第三雙音多頻訊號,且反覆偵測的次數由該話機之一每秒脈碼調變取樣頻率、該最大偵測長度及一延遲範圍常數所決定。The video conferencing method of claim 16, wherein the audio input module further detects whether the second audio segments in the second buffer include the third dual tone multi-frequency signal, and repeatedly The number of detections is determined by one of the phone's pulse code modulation sampling frequency per second, the maximum detection length, and a delay range constant. 如申請專利範圍第16項所述之視訊會議方法,其中被設定為0值之該些第二音頻片段的時間長度由一延遲基本單位、該話機之一每秒脈碼調變取樣頻率及該最大偵測長度所決定。The video conferencing method of claim 16, wherein the length of time of the second audio segments set to a value of 0 is determined by a delay basic unit, one of the telephones, and a pulse code modulation sampling frequency per second. The maximum detection length is determined.
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