1296470 九、發明說明： 【發明所屬之技術領域】 本發明係關於網路語音裝置，尤其是有關於檢測網路語 音裝置語音品質的方法。 【先前技術】 隨著網際網路的風行與成長，各種嶄新的網路應用不斷 的出現，也對人們的生活產生許多新的變化與衝擊。在這些 嶄新的網路應用中，其中最引人注目的就是各種名為網路電 話、Voice over IP (VoIP)、IP電話的裝置、系統、或軟體。 以下，為了統一起見，統稱這些使雙方得透過網際網路交談 通話的技術或做法為「網路語音」。 網路語音之所以有逐漸取代傳統、經由公眾電話交換網 路（public switched telephone network，PSTN)通話的趨勢， 主要是由於經濟的因素。傳統的PSTN通話一般是以每次通 話受話方所在的地點與通話時間長短來計費的，而網路語音 因為網際網路已經打破了距離與國家、區域的界限，而且網 路服務業者也不針對屬於網路語音的資料流量收費，因此網 路語音的成本非常低廉。但是過去因為網路頻寬的有限、相 關技術未甄完善、協定標準的不一等，網路語音只是屬於一 些玩家之間的遊戲，還無法進入家庭、辦公室等環境成為人 們生活中可靠的通訊媒介。 近年來，這些限制網路語音成長的因素——的被解決， 1296470 而網路語音的應用也開始呈現爆炸性的成長。但是無論如 何，傳統的電話已經存在有百年以上，傳_電話就像水電 一樣已經是人們生活的一部份，一般的家戶中都至少有一台 以上的傳統電話、都有已經佈設好的内部網路、還有類似答 錄機、傳真機的裝置。所以目前市面上成長最快的是一端銜 接網路、一端銜接傳統電話的網路語音裝置。如第。圖所示， 這類網路語音裝置100具有一個網路埠11〇 (通常是屬於 RJ-45的插口），以透過一個區域網路（未標號）、以及像是 ADSL數據機12〇、Cable數據機等裝置與公眾網際網路8〇〇 連線，此外這個網路語音裝置1〇〇還具有至少一個話機埠（通 苇疋屬於RJ-11的插口）130以銜接一般的電話機14〇。使用 者可以像使用傳統電話一般，拿起聽筒、透過裝置的一個使 用界面（未圖示）或是話機的數字按鍵來「撥號」、然後等待 對方接聽。對方則透過一個和網路語音裝置1〇〇相同或相容 的裝置150來應答。網路語音裝置1〇〇還可以有另一個線路 埠160與公眾電話交換網路9〇〇連接，以便和傳統電話機一 樣發話給使用傳統電話網路的受話方。 人們的語音是連續的類比訊號，網路語音是將這個連續 的類比訊號數位化、壓縮、編碼，然後以一個個的分別的資 料封包依序傳送出去。反之，在接收端將這些資料封包依序 收集起來、解碼、然後轉為人耳可聽的連續類比訊號。在這 1296470 些轉換的過程，再加上封包透過網路傳輸所經歷的各種延遲 或者丟失等傳輸上的狀況’都會影響通話雙方所感受到的通 話品質。傳統上，這些網路語音裝置的品質，主要可以分為 界面品吳（interface quality )與線路品質（1丨狀quality )兩大 部分。前者主要是有關於網路語音裝置所產生的語音的電氣 祝號的品質’包括像疋插入相失（insertion loss)、總諧波失 真（total harmony distortion， THD )、互調失真 (inter-modulation distortion，IMD )、噪音值、訊號雜音比（S/N ratio )等。後者則和網路語音裝置承受與補償封包傳送的延 遲、封包與封包之間間隔的變化（jitter)、封包的損壞與丟失 的能力有關。 傳統上要檢測網路語音裝置這兩方面的品質通常需要購 買昂貴與專屬的設備，還需要對於待測的網路語音裝置設置 一些特殊的安排。以第la圖所示的網路語音裝置1〇〇為例， 如果要測試界面品質，如第lb圖所示，一個專屬的測試設備 170直接和網路語音裝置100的話機和線路埠130、160用 RJ_U的纜線（未標號）連接。網路語音裝置100的内部還需 要安排一個話機埠和線路埠13〇、16〇之間的迴接（1〇叩back) (圖中以虛線表示）。在進行測試時，測試設備170透過線路 璋160送出特殊的測試訊號進入網路語音裝置1〇〇，測試訊 號經由迴接、話機埠13〇回到測試設備17〇。測試設備17〇 1296470 藉著比對當初送出的、以及後來收回的測試訊號，就可以計 算出相關的界面品質參數。除了特殊測試設備的投資、還有 待測裝置的迴路的設置以外’這種測試方法退有其他的缺 點，像是測試設備一定要和待測裝置位於同一地點且相隔不 能太遠，才能將兩者連結起來。如果待測裝置位於另一地點， 那麼就會需要將測試設備搬到當地來進行測試。 如果是測試線路品質，如第lc圖所示，就更複雜了，需 要另外的特殊測試設備180、190，分別和網路語音裝置100 和150的話機埠130連接。特殊的測試設備180發出一些已 知的測試訊號，經由網路語音裝置100、公眾網際網路或是 一個模擬的網路810 (可以提供不同程度的延遲、封包丟失 率等）、網路語音裝置150到達測試設備190。然後經由比對 已知的測試訊號，就可以得到相關的品質數據。除了特殊測 試設備的投資、還有模擬網路的設置以外，這種測試方法也 有其他的缺點5像是測試設備一定要和待測裝置位於同一地 點且相隔不能太遠，才能將兩者連結起來。 【發明内容】 為了克服習知的測試網路語音裝置的缺點，本發明提出 一種測試網路語音裝置的方法，這個方法不需要購置昂貴的 特殊測試設備，也不限於一定要在被測的網路語音裝置的附 近實施，被測的網路語音裝置内部也不需要特別的硬體上的 1296470 安排，只需要簡單的連線，本發明提出的方法更可以同時用 來測試界面品質以及線路品質，而不需要準備兩套不同的測 試設備和環境。 本發明所提出的測試方法，可以軟體模組的方式在一般 的運算裝置上執行。這個運算裝置是透過一個標準的區域網 路和被測的網路語音裝置連線，然後經由這個軟體的執行來 進行測試。只要將一或多個待測的網路語音裝置連接在這個 區域網路上後，就可以以本發明提出的方法·——加以測試， 所以測試的環境以及設置非常簡單，也不需要對被測的網路 語音裝置實施什麼修改。 除了區域網路以外，本發明提出的方法也可以透過廣域 網路（像是公眾網際網路）進行，所以被測的網路語音裝置 可以是位於任何透過公眾網際網路所能達到的地點，所以本 發明可以用來對安裝在遠端客戶環境裡的網路語音裝置來進 行測試，而不需要派遣人員、攜帶裝置到客戶所在來進行測 試。 本發明所提出的方法可以適用於具有一或多個話機埠的 網路語音裝置，也可以適用於沒有、或者是有線路埠的網路 語音裝置。以最常見的具有一個話機埠以及一個線路埠的網 路語音裝置為例，只要用一個標準的RJ-11纜線將這二個埠 口串接起來，就可以進行界面品質與線路品質的相關測試， 1296470 而不需要架設兩套不同的環境。 茲配合所附圖示、實施例之詳細說明及申請專利範圍， 將上述及本發明之其他目的與優點詳述於後。然而，當可了 解所附圖示純係為解說本發明之精神而設，不當視為本發明 範疇之定義。有關本發明範疇之定義，請參照所附之申請專 利範圍。 【實施方式】 本發明提出的測試網路語音裝置的方法，可以是以一種 軟體模組的方式安裝於一般的運算裝置上執行。這個運算裝 置包括（但不限於）桌上型電腦、筆記型電腦、甚至PDA。 本發明所提出的方法也可是以韌體的方式實施於一個專屬的 獨立裝置裡。這些獨立或運算裝置的特徵是需要具有一個網 路界面，可以和待測的網路語音裝置透過一個網路相通訊。 通常這個網路界面是一個RJ-45的網路璋，然後運算裝置和 待測的網路語音裝置是透過一個有線的以太網路相連接。在 其他的實施例裡，網路界面也可以是一個光纖的界面、或是 一個支援802.11協定的無線網路。也就是說，本發明不限定 使用哪一種特定的和待測網路語音裝置連線的方式和網路， 只要能達成運算裝置和待測語音裝置連線的各種網路界面與 網路都可以。 第2a圖係依據本發明之第一實施例架構於一區域網路之 1296470 示意圖。如第2a圖所示，本發明所提出的方法是以一測試程 式210的形式安裝於運算裝置200裡。測試程式210主要包 含使用界面212、測試數據產生模組214、以及數據分析模組 216。當測試程式210被執行起來後，使用界面212容許使用 者透過運算裝置200的鍵盤、螢幕（未圖示）等輸入相關的 測試參數、啟動測試的執行、以及將測試的結果呈現出來。 運算裝置200有諸多細節被省略，例如處理器、記憶體、匯 流排、硬碟、作業系統等，因為這些細節應為此一領域具一 般技藝人士所熟習，故於此不再贅述。 運算裝置200是以一網路埠202和一有線以太區域網路 220連線。測試數據產生模組214是產生一系列特定的測試 數據封包（未圖示），經由網路埠202以及區域網路220傳送 給被測的網路語音裝置230，經由網路語音裝置230處理過 的封包會經由區域網路220以及網路埠202而由數據分析模 組216所接收並與先前送出去的一系列測試封包加以比對分 析，而得出有關網路語音裝置230的品質分析。區域網路220 也有相當的細節被省略，比如說構成區域網路220 —部份的 交換器等。 被測的網路語音裝置230具有一個線路璋232和一個話 機珲234。這是最常見的一種小型的網路語音設備。線路埠 232 —般是具有所謂FXO的界面，也就是一般話機對局端的 1296470 界面，具有接收振鈴（ring)訊號、掛斷（〇n_h〇〇k)、接聽 (off-hook)、發出DTMF的按鍵音的功能，是用來和公眾電 6舌义換網路連線的界面。話機埠234 —般是具有所謂FX§的 界面，也就疋局端對話機的界面，具有提供迴路電流（1〇叩 current )、發出振聆訊號、提供撥號音（did t〇ne )、偵測掛斷 與接聽的功能，是用來和話機連線的界面。 這裡請注意到，一般網路語音裝置23〇的線路埠232具 有兩種作用，-是容許-個連接於話機埠W的話機直接發 話到線路埠232所連接的公眾電話交換網路，另-是容許從 網路埠236進來的-通網路語音來話轉接到線路埠m所連 接的公眾電話交換網路。對於後者，網路語音來話的最早的 封包裡會記錄有所要撥號到公眾電話交換網路的受話方電話 號碼，網路語音裝置23G㈣這個封包後，就會透過線路璋 232的削界面，模擬一般電話拿起話筒（off-hook)、偵測 撥號音、發域表受話方電話號碼的按鍵音、錢將受話方 和發話方經過編碼解碼模組238連結起來。另一方面，經由 車 t °舌到網際網路的另一個相同或相容裝置的方式 田^機阜234所連接的電話話筒被拿起時（off-hook)， I隼234的FXS界面偵測到這個狀況後提供撥號音給話機 斤連接的兒話，這個時候發話人可以直接用電話的按 鍵輸入對方相同或相容裝置的網路地址，網路語音裝置230 12 1296470 然後會將發話的電話、經由話機埠234、編碼解碼模組238， 和對方的裝置連結起來。 如第2a圖所示，線路埠232和話機埠234之間是以一條 RJ11纜線240連接。這樣一個簡單的連線再加上FXS與FXO 界面的特性就完成了一個簡單的迴接（loop back )的連線。 在這樣的迴接下，對線路埠232而言，話機琿234扮演了公 眾電話交換網路的角色，而對話機埠234而言，線路埠232 扮演了話機的角色。 本發明提出的測試方法運作方式說明如下。首先，測試 數據產生模組214產生的測試封包裡註明了要透過線路埠 232發話給公眾電話交換網路的某個「電話號碼」，這個「電 話號碼」其實就是運算裝置200的網路地址。這個測試封包 然後被發送給網路語音裝置230。網路語音裝置230收到這 個封包後，依照前述FXO界面的流程，先是模擬一般電話拿 起話筒（提機、off-hook)，經由迴接的線路240，話機琿234 偵測到提機，於是發出撥號音，線路埠232偵測到撥號音後， 將運算裝置200的網路地址的「電話號碼」用DTMF的按鍵 音發出，話機埠234收到這個網路地址後，即試圖與這個網 路地址所在的運算裝置200建立起網路語音通話的管道。至 此，網路運算裝置200、與網路語音裝置即構成了一個如第 2a圖虛線所示的測試迴路。 13 1296470 接下來’測試數據產生模組214繼續產生與送出特定的 測试數據封包，這些封包經由纽所㈣路徑，先被編瑪解 碼模組238解碼成類比的訊號，經由線路埠232、以及話機 璋234’再被編碼解碼模組238編碼成數據封包，經過區域 網路22G而由數據分析模組216所接收。這些封包然後和當 初發达的測4封包比對、分析，就可以得出界面與線路相關 的时貝刀析出來。μ注意到’在這個測試的過程中，網路語 音裂置23G與運算裝置2⑽的網路地址必須都是已知的。而 且上述FXO、FXS的界面功能均係—般網路語音裝置所 具備的’不需要額外的設定妹硬體的修改。另外在數據分 析方面可以用任何習知的技術或演算法來進行，因為演算 法的内容非本發_神所在，所以不讀述於此。 從以上u月可以清楚得知，本發明湘網路語音裝置 本身具備之界面與功能，以及非常簡單的連線方式，即可對 於網路語音裝置的界面品質與線路品質予以檢測。對於1他 待測的網路語音裝置，只要知道其網路位址，然後依財式 連接’重新執制試程式即可，既不f要購w特殊測試設備’ 測試環境的架設也非常簡單。 當網路語音裝置230提供-個以上的話機埠234時，可 以採用如第2b圖所示的連線方式，對於每—個話機璋w， 一—予以測試。當網路語音裝置只有—個話機埠234時，則 14 1296470 可以採用如第2c圖所示的連線方式，利用另外一台具有FXO 界面的網路語音裝置250來搭配測試。 基於同樣的架構，本發明也可以用公眾網際網路800來 取代區域網路220，如第2d圖所示。在這樣的架構下，本發 明可以透過公眾網際網路對於遠端的網路語音裝置予以檢 測。 以上的實施例都需要利用到一個FXO的界面（不論是直 接位於被測的網路語音裝置、還是利用另外一台網路語音裝 置的FXO界面），對於不具有FXO界面的網路語音裝置，或 是無法取得另外一台具有FXO界面的網路語音裝置時，本發 明可以透過如第3圖所示的一個耦合裝置來測試網路語音裝 置的FXS界面。 如第3圖所示，耦合裝置300係包含一有被動元件如電 感和電容等（未標號）所構成的簡單電路。當被測的網路語 音裝置310的話機埠312的FXS界面中習稱為Tip和Ring的 兩條導線322、324，以及話機埠314的FXS界面中Tip和 Ring導線332、334,分別如第3圖所示的方式和耦合裝置300 連接後，會導致話機埠312有迴路電流流經導線322和324、 以及話機埠314有迴路電流流經導線332和334。換言之， 接上耦合裝置300後，會使得話機埠312和314都被提機， 而耦合裝置300裡的電感會使得兩邊導線上的語音的電氣訊 15 1296470 號得以從一側耦合到另外一側，而使得話機埠312和314可 以直接交談。然後運算裝置（未圖示）可以分別對話機埠312 和314發出封包，首先要求兩個話機埠都停止發出撥號音（在 一些實施例中，如果可以將撥號音等過濾的話，這個步驟可 以被省略）。請注意到，封包裡有欄位可以註明是給哪一個話 機埠的，而測試程式（未圖示）也需要被告之目前測試的是 哪兩個話機埠。接著測試程式的測試數據產生模組可以針對 其中任一個話機埠發出測試數據封包，以話機埠312為例， 這些封包經由編碼解碼模組（未圖示）解碼成類比的訊號， 經由話機埠312、耦合裝置300、話機埠314，再被編碼解碼 模組編碼成數據封包，經過網路而由測試程式的數據分析模 組所接收。這些封包然後和當初發送的測試封包比對、分析， 就可以得出界面與線路相關的品質分析出來。除了適用於具 有多個話機埠的網路語音裝置外，本發明可以將兩個網路語 音裝置的話機埠依照第3圖所示的方式連線起來進行測試。 同樣的，第3圖所示的測試方式也都適用於區域網路或廣域 網路的環境。 藉由以上較佳具體實施例之詳述，係希望能更加清楚描 述本創作之特徵與精神，而並非以上述所揭露的較佳具體實 施例來對本創作之範疇加以限制。相反地，其目的是希望能 涵蓋各種改變及具相等性的安排於本創作所欲申請之專利範 16 1296470 圍的範疇内。 【圖式簡單說明】 第la圖係習知之網路語音裝置之連線示意圖。 第lb圖係習知之網路語音裝置測試界面品質之連線示意圖。 第lc圖係習知之網路語音裝置測試線路品質之連線示意圖。 第2a圖係依據本發明之第一實施例架構於一區域網路之示意 圖。 第2b圖係依據本發明之第二實施例架構於一區域網路之示 意圖。 第2c圖係依據本發明之第三實施例架構於一區域網路之示意 圖。 第2d圖係依據本發明之第四實施例架構於一廣域網路之示 意圖。 第3圖係依據本發明之第五實施例之一耦合裝置之示意圖。 【主要元件符號說明】 100 網路語音裝置 110 網路埠 120 ADSL數據機 130 話機埠 140 電話機 150 網路語音裝置 160 線路埠 170 測試設備 180 測試設備 190 測試設備 200 網路語音裝置 202 網路埠 17 測試程式 測試數據產生模組 有線以太區域網路 線路埠 網路埠 RJ11欖線 耦合裝置 話機埠 Tip導線 Tip導線 公眾網際網路 公眾電話交換網路 212 使用界面 216 數據分析模組 230 網路語音裝置 234 話機埠 238 編碼解碼模組 250 網路語音裝置 310 網路語音裝置 314 話機埠 324 Ring 導線 334 Ring 導線 810 模擬網路 181296470 IX. Description of the Invention: [Technical Field] The present invention relates to a network voice device, and more particularly to a method for detecting voice quality of a network voice device. [Prior Art] With the popularity and growth of the Internet, various new network applications have emerged, and many new changes and impacts have been made on people's lives. Among the new web applications, the most notable are devices, systems, or software called network over IP (VoIP), IP telephony. In the following, for the sake of unification, these technologies or practices that enable both parties to talk over the Internet are called "network voice." The trend of VoIP has gradually replaced the traditional, public switched telephone network (PSTN) calls, mainly due to economic factors. The traditional PSTN call is generally charged by the location of the caller and the duration of the call, and the voice of the Internet has broken the distance from the country and the region, and the network service provider does not. The charging of data traffic belonging to VoIP is very low, so the cost of VoIP is very low. However, in the past, because of the limited network bandwidth, the lack of relevant technologies, and the different standards of the agreement, VoIP is only a game between some players, and it is impossible to enter the home, office and other environments to become reliable communication in people's lives. medium. In recent years, these factors that limit the growth of VoIP have been solved, 1296470, and the application of VoIP has begun to show explosive growth. But in any case, traditional telephones have existed for more than a hundred years. The telephone is like a hydropower. It is already part of people's lives. In general, there are at least one traditional telephone and all the interiors that have been laid out. Internet, and devices like answering machines and fax machines. Therefore, the fastest growing market is the VoIP device that connects to the Internet at one end and connects to traditional phones at one end. As the first. As shown, such a network voice device 100 has a network port 11 (usually a jack belonging to the RJ-45) for transmission through a regional network (not labeled), and such as an ADSL modem 12, Cable. A device such as a data machine is connected to the public internet network. In addition, the network voice device 1 has at least one telephone port (through the jack belonging to the RJ-11) 130 to connect the general telephone unit 14A. The user can pick up the handset, use a user interface (not shown) or the phone's numeric buttons to "dial" like a traditional telephone, and then wait for the other party to answer. The other party responds via a device 150 that is identical or compatible with the network voice device. The VoIP device 1 can also have another line 埠 160 connected to the public switched telephone network 9 发 to communicate with the traditional telephone set to the caller using the conventional telephone network. People's voice is a continuous analog signal. Network voice digitizes, compresses, and encodes this continuous analog signal, and then transmits it in sequence by separate data packets. Conversely, at the receiving end, these data packets are collected, decoded, and then converted into continuous analog signals that are audible to the human ear. In the process of these 1296470 conversions, and the various delays or loss of transmissions experienced by the packets transmitted through the network, the quality of the calls perceived by both parties will be affected. Traditionally, the quality of these VoIP devices can be divided into two parts: interface quality and line quality. The former is mainly about the quality of the electrical designation of the voice generated by the VoIP device, including the insertion loss, the total harmonic distortion (THD), and the inter-modulation. Distortion, IMD), noise value, signal noise ratio (S/N ratio), etc. The latter is related to the ability of the VoIP device to withstand the delay of packet transmission, the jitter between packets and packets, and the ability to corrupt and lose packets. Traditionally, to detect the quality of both aspects of VoIP devices, it is often necessary to purchase expensive and proprietary devices, as well as to set up special arrangements for the VoIP devices to be tested. Taking the network voice device 1 shown in FIG. 1 as an example, if the interface quality is to be tested, as shown in FIG. 1b, a dedicated test device 170 directly connects to the phone and line 130 of the network voice device 100. 160 is connected by a cable (not labeled) of RJ_U. The inside of the VoIP device 100 also needs to arrange a callback between the telephone 埠 and the lines 〇13〇, 16〇 (indicated by a broken line in the figure). During the test, the test device 170 sends a special test signal through the line 璋 160 to the VoIP device 1 , and the test signal is returned to the test device 17 via the call back and the phone 埠 13 . The test equipment 17〇 1296470 can calculate the relevant interface quality parameters by comparing the test signals sent out and later recovered. In addition to the investment in special test equipment and the setting of the circuit of the device under test, this test method has other disadvantages. For example, the test equipment must be co-located with the device under test and not too far apart to Connected together. If the device under test is located at another location, then the test equipment will need to be moved to the local area for testing. If the test line quality, as shown in Figure lc, is more complicated, additional special test equipment 180, 190 is needed to connect to the telephone 埠 130 of the network voice devices 100 and 150, respectively. The special test equipment 180 sends out some known test signals via the VoIP device 100, the public internet or an analog network 810 (which can provide different degrees of delay, packet loss rate, etc.), VoIP devices. 150 arrives at test device 190. The relevant quality data can then be obtained by comparing the known test signals. In addition to the investment in special test equipment and the setup of analog networks, this test method has other shortcomings. 5 The test equipment must be co-located with the device under test and not too far apart to connect the two. . SUMMARY OF THE INVENTION In order to overcome the shortcomings of the conventional test network voice device, the present invention provides a method for testing a network voice device, which does not require the purchase of expensive special test equipment, nor is it necessarily limited to the network to be tested. In the vicinity of the voice device, the internal network voice device does not need a special hardware 1296470 arrangement, and only requires a simple connection. The method proposed by the present invention can simultaneously be used to test interface quality and channel quality. Instead of preparing two different sets of test equipment and environments. The test method proposed by the present invention can be executed on a general arithmetic device in the form of a software module. The computing device is connected to the measured VoIP device through a standard regional network and then tested by the execution of the software. As long as one or more network voice devices to be tested are connected to the local area network, the method proposed by the present invention can be tested, so the test environment and the setting are very simple, and the test is not required. What modifications are implemented by the VoIP device. In addition to the local area network, the method proposed by the present invention can also be performed through a wide area network (such as the public internet), so the measured VoIP device can be located at any location accessible through the public internet, so The present invention can be used to test VoIP devices installed in a remote client environment without the need to dispatch personnel or carry devices to the customer for testing. The method proposed by the present invention can be applied to a network voice device having one or more telephones, and can also be applied to a network voice device without or having a line. Taking the most common network voice device with one phone and one line, as long as the two ports are connected in series by a standard RJ-11 cable, the interface quality can be related to the line quality. Test, 1296470 without the need to set up two different environments. The above and other objects and advantages of the present invention will be described in detail with reference to the accompanying drawings and claims. However, it is to be understood that the appended drawings are merely illustrative of the scope of the invention. For the definition of the scope of the invention, please refer to the attached patent application. [Embodiment] The method for testing a network voice device proposed by the present invention may be installed on a general arithmetic device in a manner of a software module. This computing device includes, but is not limited to, a desktop computer, a notebook computer, or even a PDA. The method proposed by the present invention can also be implemented in a proprietary, stand-alone device in a firmware. These independent or computing devices are characterized by the need to have a network interface that can communicate with the network voice device to be tested over a network. Usually the network interface is an RJ-45 network port, and then the computing device and the network voice device to be tested are connected through a wired Ethernet channel. In other embodiments, the network interface can also be a fiber optic interface or a wireless network that supports the 802.11 protocol. That is to say, the present invention does not limit which specific method and network to use to connect to the VoIP device to be tested, as long as the various network interfaces and networks connecting the computing device and the voice device to be tested can be achieved. . Figure 2a is a schematic diagram of 1296470 architecture of a regional network in accordance with a first embodiment of the present invention. As shown in Fig. 2a, the method proposed by the present invention is installed in the arithmetic unit 200 in the form of a test procedure 210. The test program 210 mainly includes a usage interface 212, a test data generation module 214, and a data analysis module 216. After the test program 210 is executed, the usage interface 212 allows the user to input relevant test parameters, initiate test execution, and present the test results through the keyboard, screen (not shown), etc. of the computing device 200. The details of the computing device 200 are omitted, such as processors, memory, bus, hard disk, operating system, etc., as such details should be familiar to those skilled in the art and will not be described again. The computing device 200 is connected by a network port 202 and a wired Ethernet area network 220. The test data generation module 214 generates a series of specific test data packets (not shown), which are transmitted to the measured network voice device 230 via the network 202 and the regional network 220, and processed by the network voice device 230. The packet is received by the data analysis module 216 via the local area network 220 and the network port 202 and compared with a series of test packets sent previously, and the quality analysis of the network voice device 230 is obtained. The local area network 220 also has considerable details that are omitted, such as a switch that forms part of the local area network 220. The VoIP device 230 under test has a line 璋 232 and a phone 234. This is the most common type of small network voice device. Line 埠 232 is generally an interface with so-called FXO, which is the 1296470 interface of the general telephone to the central office, with receiving ring signal, hang up (〇n_h〇〇k), take-off (off-hook), issue DTMF The function of the button sound is used to interface with the public electricity. The phone 埠234 is generally an interface with the so-called FX§, which is also the interface of the central office dialogue machine, which provides loop current (1〇叩current), emits a vibration signal, provides dial tone (did t〇ne), and detects The function of measuring the hang up and answer is the interface used to connect with the phone. Please note here that the general network voice device 23 〇 line 232 has two functions, that is, allows a telephone connected to the phone 直接W to directly send a call to the public telephone switching network connected to the line 232, and It is to allow incoming voice over the network 236 to be transferred to the public switched telephone network connected to the line 埠m. In the latter case, the earliest packet of the incoming voice of the network voice will record the telephone number of the called party to be dialed to the public switched telephone network. After the packet of the network voice device 23G (4), it will be simulated through the interface of the line 232. Generally, the phone picks up the microphone (off-hook), detects the dial tone, and dials the button number of the telephone number of the recipient's party. The money connects the callee and the caller through the codec module 238. On the other hand, the phone microphone connected to the phone 234 via the t° tongue to the other identical or compatible device of the Internet is off-hook, IFX234 FXS interface detection After detecting this condition, the dial tone is provided to the phone to connect the phone. At this time, the caller can directly input the network address of the same or compatible device by using the button of the phone, and the network voice device 230 12 1296470 will then send the message. The telephone, the telephone 234, and the codec module 238 are connected to the other party's device. As shown in Figure 2a, the line 埠 232 and the phone 埠 234 are connected by an RJ11 cable 240. This simple connection, combined with the features of the FXS and FXO interfaces, completes a simple loop back connection. Under such a connection, for line 232, phone 234 acts as a public switched telephone network, and for port 234, line 232 acts as a telephone. The mode of operation of the test method proposed by the present invention is explained below. First, the test packet generated by the test data generation module 214 indicates a "telephone number" to be sent to the public switched telephone network via the line 232. This "telephone number" is actually the network address of the computing device 200. This test packet is then sent to the network voice device 230. After receiving the packet, the VoIP device 230 firstly simulates the general phone to pick up the microphone (lifting, off-hook) according to the flow of the FXO interface, and the phone 珲 234 detects the lifting device via the returned line 240. Then, a dial tone is issued, and after the line 埠 232 detects the dial tone, the "phone number" of the network address of the computing device 200 is sent with the DTMF button tone, and the phone 埠 234 receives the network address, and then attempts to The computing device 200 where the network address is located establishes a conduit for voice over internet calls. Thus, the network computing device 200 and the network voice device constitute a test circuit as indicated by the broken line in Fig. 2a. 13 1296470 Next, the test data generation module 214 continues to generate and send specific test data packets, which are decoded into analog signals by the codec module 238 via the button (4) path, via the line 232, and The phone 璋 234 ′ is further encoded into a data packet by the code decoding module 238 and received by the data analysis module 216 via the local area network 22G. These packets are then compared and analyzed with the newly developed test 4 packets, and the interface-line-related time-knife can be obtained. μ noted that during the course of this test, the network address of the network burst 23G and the computing device 2 (10) must be known. Moreover, the interface functions of the above FXO and FXS are all provided by the general network voice device, and no additional configuration hardware modification is required. In addition, data analysis can be performed by any conventional technique or algorithm, because the content of the algorithm is not the same as the present, so it is not described here. It can be clearly seen from the above u month that the interface and function of the Xiang network voice device of the present invention and the very simple connection method can detect the interface quality and the line quality of the network voice device. For 1 VoIP device to be tested, as long as you know its network address, then connect to 're-execute the test program according to the financial system, neither f want to buy w special test equipment' test environment is very simple to set up . When the network voice device 230 provides more than one phone 234, it can be tested for each phone 璋w, one by using the connection method as shown in Fig. 2b. When the network voice device has only one phone 234, the 14 1296470 can be connected to the test by using another network voice device 250 with an FXO interface as shown in Figure 2c. Based on the same architecture, the present invention can also replace the regional network 220 with the public internet 800, as shown in Figure 2d. Under such an architecture, the present invention can detect remote VoIP devices over the public Internet. All of the above embodiments need to utilize an FXO interface (whether directly on the VoIP device under test or FXO interface using another VoIP device). For VoIP devices that do not have an FXO interface, Alternatively, when another VoIP device having an FXO interface is not available, the present invention can test the FXS interface of the VoIP device through a coupling device as shown in FIG. As shown in Fig. 3, the coupling device 300 includes a simple circuit composed of passive components such as inductance and capacitance (not labeled). When the FXS interface of the phone 埠 312 of the measured VoIP device 310 is referred to as two wires 322, 324 of Tip and Ring, and the FXS interface of the phone 314, the Tip and Ring wires 332, 334 are respectively The manner shown in Figure 3 is coupled to the coupling device 300, which causes the handset 312 to have loop current flowing through the conductors 322 and 324, and the handset 314 to have loop current flowing through the conductors 332 and 334. In other words, after the coupling device 300 is connected, the telephones 312 and 314 are both lifted, and the inductance in the coupling device 300 causes the electrical signal 15 1296470 of the voice on the two sides to be coupled from one side to the other. So that the phones 312 and 314 can talk directly. The computing device (not shown) can then issue a packet to each of the talk boxes 312 and 314, first requiring both phones to stop issuing a dial tone (in some embodiments, if a dial tone or the like can be filtered, this step can be Omitted). Please note that there is a field in the packet that indicates which phone is being given, and the test program (not shown) also needs to be told which two phones are currently being tested. Then, the test data generation module of the test program can send test data packets to any of the phones, taking the phone 312 as an example. The packets are decoded into analog signals via a codec module (not shown), via the phone 312. The coupling device 300 and the phone 314 are encoded into data packets by the codec module, and are received by the data analysis module of the test program via the network. These packets are then compared and analyzed with the test packets sent at the beginning, and the quality analysis of the interface and the line can be obtained. In addition to being applicable to a network voice device having a plurality of telephones, the present invention can test the telephones of two network voice devices by connecting them in the manner shown in FIG. Similarly, the test methods shown in Figure 3 are also applicable to regional or wide area network environments. The features and spirit of the present invention are more clearly described in the above detailed description of the preferred embodiments, and the scope of the present invention is not limited by the preferred embodiments disclosed herein. On the contrary, the purpose is to cover a variety of changes and equivalence arrangements within the scope of the patent application 16 1296470 to which the present application is intended. [Simple description of the diagram] The first diagram is a schematic diagram of the connection of the conventional network voice device. Figure lb is a schematic diagram of the connection quality of the network voice device test interface. The lc diagram is a schematic diagram of the connection of the quality of the network voice device test circuit. Figure 2a is a schematic diagram of a network architecture in accordance with a first embodiment of the present invention. Figure 2b is an illustration of the architecture of a regional network in accordance with a second embodiment of the present invention. Figure 2c is a schematic diagram of a network architecture in accordance with a third embodiment of the present invention. Figure 2d is an illustration of the architecture of a wide area network in accordance with a fourth embodiment of the present invention. Figure 3 is a schematic view of a coupling device in accordance with a fifth embodiment of the present invention. [Main component symbol description] 100 network voice device 110 network 埠 120 ADSL data machine 130 phone 埠 140 telephone 150 network voice device 160 line 埠 170 test device 180 test device 190 test device 200 network voice device 202 network 埠17 Test program test data generation module wired Ethernet network line network 埠 RJ11 lanyard coupling device phone 埠 Tip wire Tip wire public internet public telephone exchange network 212 use interface 216 data analysis module 230 network voice Device 234 Phone 埠 238 Codec Module 250 Network Voice Device 310 Network Voice Device 314 Phone 324 Ring Wire 334 Ring Wire 810 Analog Network 18