TW466864B - System for multiple voice lines with data over a single subscriber loop - Google Patents
System for multiple voice lines with data over a single subscriber loop Download PDFInfo
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- TW466864B TW466864B TW89106612A TW89106612A TW466864B TW 466864 B TW466864 B TW 466864B TW 89106612 A TW89106612 A TW 89106612A TW 89106612 A TW89106612 A TW 89106612A TW 466864 B TW466864 B TW 466864B
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A7 466864 ______B7_______ 五、發明說明(1 ) 發明領域- 本發明與數位用户迴路應用有關’特別是與單一共享用 户迴路之多重數據語音線路有關。 發明背景· 隨著網際網路的普及與在家小企業的漸增,電話服務提 供業者日益受到企業與住家欲另增語音線路服務的強烈要 求。 大部份之中央機房具有提供额外之語音線路至用户端之 額外交換容量。一旦額外的接取線路延伸到了用户端,電 話服務提供業者僅需花很小的費用於提供語音服務,但增 加的線路所能提供的服務卻可而讓電話服務提供業者增加 利潤。 傳統提供額外接取線路的方法是,再鋪設銅線至用户 端,並新增或變更用户端閃電保護裝置以增加類比用户迴 予。所謂用户迴路是電話用户終端設備與服務中央機房 (或其他終端設備)間之雙銅線傳輸路徑與信令路徑。然 而,此方法所涉入的時間與費用是很巨大的,而耗費時間 會降低投資利澗。 從傳輸的觀點來看,任何類比式用户迴路爲基礎之信號 系統都存在著信號遺失及信號損害的問題。這樣的情況乃 是因實體條件所導致,像是橋式分接頭,線徑的改變,線 路的長度,絕緣,老化以及環境纜線的損害,或由外來之 像是,脈衝雜訊以及_音的干擾所造成。所謂信號的衰 退,一般來講就是雜訊、遺失、失眞以及干擾。 -4 - 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) 、---^--------'.裝 i — (请先閱讀背面之it意事項再填寫夫頁) 訂 --線. 經濟部智慧財產局員工消費合作社印製 經濟部智莨財產局員工消費合作社印製 4 S 6 8 6 4 A7 _— _ _ B7 五、發明說明(2 ) 使用傳統方法的另一個問題是,類比迴路一般所使用的 都是標準數據機,而標準數據機乃是利用基頻POTS (基本 電話服務)語音頻譜(0 - 4讦赫茲)來傳送資訊,由於缆線 對的_音效應,傳送功率不可以超過聯邦通信委員會所规 定的等級。現今聯邦通信委員會規定服務提供業者數據機 的輸出下傳速度應限制在每秒5萬3仟個位元,上傳速度 則限制在每秒3萬1什2佰個位元。實際的速度可能會隨著 線路條件的不同而不同,但不能超過此上限。 分頻多工(FDM)是能在用户迴路上提供出額外語音線 路,但又不需另鋪設銅線的一種技術。此方法所使用的頻 譜與基頻POTS所使用的頻譜是有所區隔的,這使得我們 可以在同一條的雙線用户迴路上,讓另增的4仟赫茲類比 POTS通道使用較高頻的載波。但這種帶頻類比載波技術 卻會讓類比迴路遭受較傳統方式更多的信號遺失與更大的 信號損害。 數位線路倍增(DAML)就是一種屬於FDM的技術。在局 ,y 端,一個DAML數據機將供應給兩或多個用户迴路數位信 號來使用。該數據機將這些類比語音信號轉換成數位線路 碼格式,然後透過單一用户迴路將其傳送至另一位於用户 端(或用户端附近)之DAML數據機。用户端之DAML數據 機會將該線路碼予以解碼,爲用户提供出相應於局端 DAML數據機所連結之用户迴路的兩或多個雙線連結。有 許多種形式的碼可作爲該數位線路碼.,最常用的形式有振 幅、相位與頻率鍵移,2 -二位元-1-回位元,無載波振幅A7 466864 ______B7_______ V. Description of the Invention (1) Field of the Invention-The present invention is related to digital subscriber loop applications', especially to multiple data voice lines sharing a single subscriber loop. BACKGROUND OF THE INVENTION · With the popularity of the Internet and the growing number of small businesses at home, telephone service providers are increasingly demanded by businesses and homes to add voice line services. Most central computer rooms have additional switching capacity that provides additional voice lines to the client. Once the additional access line is extended to the subscriber, the telephone service provider only needs to pay a small fee to provide voice services, but the services provided by the additional line can increase the profit of the telephone service provider. The traditional method of providing additional access lines is to lay copper wires to the client and add or change the lightning protection device at the client to increase the analog user response. The so-called user circuit is the dual copper wire transmission path and signaling path between the telephone user terminal equipment and the service central computer room (or other terminal equipment). However, the time and cost involved in this method is huge, and the time consumption will reduce the investment profit. From a transmission point of view, any analog user loop-based signal system has problems with signal loss and signal damage. This situation is caused by physical conditions, such as bridge taps, changes in wire diameter, line length, insulation, aging, and damage to environmental cables, or from external sources such as pulse noise and audio Caused by interference. The so-called signal decay is generally noise, loss, loss, and interference. -4-This paper size is in accordance with China National Standard (CNS) A4 (210 X 297 mm), --- ^ -------- '. Install i — (Please read the Italian notice on the back before (Fill in the husband's page) Order--line. Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs and printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs and printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs. Another problem with the traditional method is that analog modems generally use standard modems, and standard modems use the baseband POTS (basic telephone service) voice spectrum (0-4 Hz) to transmit information. The sound effect and transmission power of the pair must not exceed the level specified by the Federal Communications Commission. At present, the Federal Communications Commission stipulates that the output speed of the service provider's modem should be limited to 53,000 bits per second, and the upload speed should be limited to 31,200 bits per second. Actual speed may vary depending on line conditions, but this limit cannot be exceeded. Frequency division multiplexing (FDM) is a technology that can provide additional voice lines on the user circuit without the need for additional copper wires. The spectrum used by this method is separated from the spectrum used by the fundamental frequency POTS, which allows us to use the higher frequency of the additional 4 仟 Hz analogy POTS channel on the same two-line user loop. Carrier. However, this band-frequency analog carrier technology will cause the analog loop to suffer more signal loss and greater signal damage than traditional methods. Digital line doubling (DAML) is a technology that belongs to FDM. At the office and y-side, a DAML modem will supply two or more user loop digital signals for use. The modem converts these analog voice signals into a digital line code format, and then transmits it through a single user loop to another DAML modem located at the client (or near the client). The DAML data of the client will decode the line code to provide the user with two or more two-line connections corresponding to the user loop connected to the central DAML modem. There are many forms of codes that can be used as this digital line code. The most commonly used forms are amplitude, phase, and frequency key shifts, 2-2 bits-1 bit, and no carrier amplitude.
I -5- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) (請先閱讀背面之法意事項再填寫本頁) 裝 *-----1— 訂·---·線 466864 A7 五、發明說明(3 ) 與相位調變以及正交振幅相位調變。此種方法的問題在 於V在局端所執行的由該博碼調變(pcM)數位信號轉換 成類比迴路信號,以及再由該類比迴路信號轉換回該 DAML數位信號之數位/類比/數位轉換,會產生量化誤差 與相位失眞而導致信號品質的下降。 另-個以數位方式在用户迴路上傳輸多重語音線路的技 術是整體服務數位網路(ISDN)。它是—個虛接數位的多 重語音/數據通道並包含信號通道的系統。不過,13〇1^需 要更改交換系統設備’以及交換機系統的管理與維護方 式。 另一種方法是在含有用户迴路的數據網路上傳送語音封 包。此法中較爲人所知的是,在網際網路協定上傳送語音 (VOIP ),在非同步傳輸模式上傳送語音(v〇ATM)以及在 訊框轉送上傳送語音(VOFR)。 VOIP —般是應用在校園環境,其每一個終端機所使用 的線材均爲電子工業聯盟/電信工業協會於1991年6月所 公佈的EIA/TIA-570-91標準:”住家與輕商業電jj線材”中 所描述的種類5線材或光纖;每一個終端機均連往一通用 殳換結構’像是乙太網路、非同步傳輸模式或混合系統。 另外’透過像是路由器這樣的閘道或是層3交換系統,可 呼叫該校園網路橋接至網際網路或是企業内網路。 在某些場合中,桌上型電腦和其他的裝置可作爲VOIP 的致能終端機,我們使用致能終端機來支援符合國際電信 聯盟(ITU)於1998年2月所公佈的ITU-T標準H.323 "封包塑 -6- 本紙張尺度通用中國國家標準(CNS)A4規格(21〇 x 2g7公釐) ί請先閲婧背面之注意事項再填寫本頁) 裝--------訂----^-----線· $濟部智慧財產局員工消費合作社印製 經濟部智慧財產局員工消費合作社印製 466864 A7 · ' :------- 五、發明說明(4 ) 多媒體通訊系統"之遠端通訊。這種系統在桌上電腦端一 般是使用數位信號處理器(DSPs)來執行語音”封包的壓 縮,然後再將它們饋送至區域中企業内網路上之其他工作 站或是透過網際網路傳送至遠端的工作站。v〇atm與 VOFR則是另種利用公眾交換電話網路(pSTN)來傳輸語音 及網接之分封技術。 用户迴路可以使用數位用户迴路系列之信號與傳輸技 術,延伸到廣域網路來作V0IP應用。數位用户迴路系列 的技術,其所送至用户端之頻寬可與基頻卩〇以服務共同 存在。ISDN也可以提供頻寬至與分封網路有連接之住宅 中,透過分封網路!^!^也可提供出語音服務。ISDN或是 數位用户迴路系列之技術(譬如,非對稱數位用户迴路與 歐規高速數位用户迴路)也可以在用户迴路上傳輸1?封 包,ATM細胞或其他的訊框格式。 不過,該網路服務而言,語音與數據的基本要求是不同 的。語音傳送僅需小量的頻寬,但所供應出來的頻寬基本 上必須疋專用或連續的且具非常小之延遲、延遲變異或遺 失。即便是延遲在毫秒的範圍内,也會在對話中產生明顯' 的回音或間斷。譬如,因路由器及閘道所引起的延遲就會 對語音產生不良的影響。 封包化的語音屬於即時交通數據的範疇,因此有關遺失 與錯誤方面的傳遞基本要求就很嚴格。在封包化的語音 中,網路端對端的平均傳遞時間必須要很小,且該端對端 傳遞時間的變化’包括遗失問題,也必須要很小。 -7- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) II 一 '裝 訂---------線· (請先閲讀背面之法意事項再填寫本頁) 4 6 6 86 4 'Μ'濟部智慧財產局員工消費合作社印製 A7 B7 五、發明說明(5 ) 在語音傳輸中,整體的延遲不應超過2()〇毫秒,此爲商 業化所能接受的數値。100-200毫秒是一般所追求的目 標。若延遲是在8 〇 〇毫秒左右,就會妨礙到正常的通話。 正常狀沉下’對於200-800毫秒的延遲,若是其不常發生 且發生在短通話時,則是可以有條件接受的。 在傳統的語音網路中,往復延遲時間大約是2〇_3〇毫 移。訊框轉送網路中之語音延遲則可以是125-20,0毫秒左 右。乙太網路承載TCP/IP封包中的延遲時間則隨訊務'量的 情況而定’其變動範園較廣β由於共享數據網路技術在本 貧上缺足即時性’所以,上述的延遲要求對要以延伸至用 户端之一般校園網路環境來傳送語音而言,是很嚴苛的挑 戰。 另外,作爲標準之ATM仍然缺乏對語音壓縮、靜音抑 制、閒置通道細胞的抑制之支援,以及對包括了將語音信 號轉換成交換虛擬連結ATM信號的信號支援的缺乏。 另外,由於A T Μ網路以及細胞組合/解組延遲所引入的 細胞延遲變異,會被用於窄頻服務(像是,語音服務)之 ATM中繼器緩衝累積,所以,除了 ATM網路本身所產生 的延遲之外’ ATM中繼器也會引入一些額外的延遲。 於是,提供出一種系統,其可在與POTS共享之用户迴 路上’提供出額外的語音與數據線路,是本發明之目標。 爲此系統提供出對任何支援進階電話特性之頻帶内信號 的支援性,是本發明之另一目標。 -8- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) — — — — — 1 — ---— I I 訂-----]I JN^ (請先閱讀背面之注意事項再填寫本頁> 4 6 6 864 A7 __ B7 五、發明說明(6 ) 爲此系統提供出幾乎沒有衰減之重建語音信號,是本發 明之另一目標。 爲此系統提供出語音交通之優先權高於數據交通,是本 發明之另一目標。 爲此系統提供出不超過該聯邦通訊委員會對類比線路所 要求之每秒5萬3仟3佰位元上限的線路類比數據機傳輸速 度,是本發明之另一目標。 爲此系統提供出可避免因額外數位/類比轉換而造成信 號損害的特性,是本發明之另一目標。 爲此系統提供出較低的數據網路環境中常見元件的使用 成本,是本發明之另一目標。 爲此系統提供出在用户端用户可自行裝設之特性,是本 發明之另一目標。 爲此系統提供出除一開始時提供出語音服務外,還可在 網路中漸次實施之特性,且毋需在現有的中央機房交換機 之外,另建獨立的數據傳輸底層結構。 發明摘要 本發明是一種系統’其可在單一的雙線用户迴路上提供 出額外的語音線路’ JL在此同時,該迴路上仍能保留用户 迴路的POTS服務。此系統包含中央機房(c 〇 )中之數位數 據機’其透過該用户迴路,與用户端中之另—數位數據機 連接〇該局端(中央機房)數位數據機透過直接數位介面連 接至局端交換機,再透過交換機連接至公眾交換電話網 路。此種接法去除了先前大部分的技藝系統中可見到的額 -9- 笨紙張尺度適用中國國家標準(CMS)A4規^各(210 X 297公髮) (請先閲讀背面之注意事項再填窝本頁) 裝--------訂------—1·線. 經濟部智慧財產局員工消費合作社印製 466864 A7 B7 五、發明說明(7 ) 外的數位/類比轉換。這些數位/類比轉換是信號衰弱的首 要原因’所以即便是只去除了—個這樣的轉換,也將會爲 系統提供出較先前技藝系統優良的信號。該局端數據機還 可以藉由分封介面(像是乙太網路)連接至數據網路。 局端數據機透過該數位的中繼介面,接收從局端交換機 而來之語音資料,該語音資料乃博碼調變(PCM )格式,以 網路定時參考信號爲時脈。然後,該PCM語音樣本會被 該局端數據機分封化,並使用適當的數位調變線路碼,傳 送至用户端數據機。局端數據機中有一個表格,此表格記 載著數位中繼介面時槽與用户端數據機之電話線路位址此 二者間的對應關係。該局端數據機還會將一個與網路定時 參考信號同步的定時參考信號?傳送至用户端數據機。上 行方向則是先接收從用户端數據機而來之p C Μ樣本語音 封包,然後予以解包並透過該數位中繼介面將之提供至局 端交換機’此時之語音爲PCM格式且以網路定時參考信 號爲其時脈。 在用户端,有一或多個電信裝置連接至該數位數據機中 之語音介面。這些電信裝置可以包括電話機具,像是傳統 電話、"智慧電話"、類比數據機或是複製(傳眞)機。 從局端數據機所傳送出來之PCM樣本語音封包,會爲 用户端數據機所接收\然後再以該定時參考信號作爲時 脈,將該語音封包轉換成類比語音頻帶訊號;之後再傳送 至已定址之電話設備。上行方向則是使用該定時參考信號 爲時脈,將從該電話設備出來之類比語音頻帶信號轉換成 -10- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公g ) (請先閲讀背面之注意事項再填寫本頁) 裝--------訂·--------線 經濟部智慧財產局員工消費合作社印製 466864 A7 B7 經濟部智慧財產局員工消費合作社印製 五、發明說明(8 ) PC Μ樣本;然後再將PCM樣本分封化爲語音封包;再由 用户端數據機使用數位調變線路碼,將語音封包透過用户 迴路傳送至局端數據機。 在用户端,數據裝置可以透過數據介面,與用户端數據 機連接。該數據裝置可以是任何的數據封包源,譬如,電 腦、橋接器、路由器或是後面接著許多獨立電腦的集線 器。用户端數據機所接收到的由局端數據機而來之數據封 包,會經由選徑而被送往該已定址之數據裝置。若是上行 方向,則會使用該數位調變線路碼,透過用户迴路將該由 數據裝置所產生之數據封包傳送至局端數據機a 此系統架構中唯一的一個非數位傳送的部份是,用户端 中之將電信裝置連接至用户端數據機之類比迴路。但此迴 路的距離通常只有幾呎。傳統系統中因電話設備與局端交 換機間有類比路徑存在而會有的信號損失及損害,因本架 構的數位性而大輻地降低。 在本發明中’語音與數據封包均是透過用户端迴路,傳 送至該中央機房。不過,本發明最主要的目的是要保存語 青通信的易理解性。透過語音與數據介面以及使用賦予語 音封包較數據封包爲高之優先順序權之封包傳送優先順序 法’將局端與用户端數據機中之語音與數據予以區分,此 主要目的可達成。美國專利字號5,692,〇35 t〇 〇,Mah〇ney et al ’以及坎貝爾,庫索恩’哈欽森,電腦通信評論之1994 年四月卷2,24册中"服務架構之品質"均有描述此種可賦 予某封包等級某傳送優先順序之封包型系統的例子。此封 -11 - ---』---------—— {請先閲讀背面之注意事項再填寫本頁) 訂: 線. 本紙張尺度適用中國國家標準(CNS)A4規格(210 297公釐) 466864 A7 B7 五、發明說明(9 匕傳送優先順序法,確保了在有數據存在的情況下,語音 的傳輸時間仍足夠的低,因而確保了語音的易理解性,另 外,此法也克服了 V0IP,VOATM以及V0FR中會碰到的 語音易理解性的問題。 圖式之描述 圖1是本發明較佳具體實施之方塊圖。 圖2是本發明獲取語音平台之方塊圖。 圖3是本發明之獲取語音平台中之語音處理卡的方塊 圖。 圖4是本發明之獲取譁音平台中之資料處理卡的方塊 圖0 圖5是本發明之獲取語音平台中之線路卡的方塊圖。 .圖6是本發明之獲取語音數據機的方塊圖。 圖7是本發明之獲取語音平台中語音處理卡方塊圖的第 二具體實施例》 圖8是本發明之獲取語音平台中資料處理卡方塊圖的第 二具體實施例。 較佳具體實施之説明 圖1是本發明之較佳具體實施的方塊圖,爲方便説明, 此圖中之中央機房僅連接單一個用户。一般其實會連接多 個用户。在局端,獲取語音平台101以绞線對連結1〇3連 接至用户迴路102。獲取語音平台ιοί另以直接數位pcM 中繼連結1 0 5連接至局端交換機1 〇 4,此中繼連結使用的 是北美標準8位元非壓縮的律集縮p c Μ技術;此外,獲 -12- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公楚) (請先閲讀背面之注意事項再填寫本頁) 装--------訂----- 線 經濟部智慧財產局員工消費合作社印製 4 6 6 8 6 4 經濟部智慧財產局員工消費合作社印製 A7 B7 五、發明說明(1〇 ) 取語音平台101還以連結106連接至數據網路。局端交換 機104則是連接至用户迴路1〇2,其也透過中繼連結107 連接至公眾交換電話網路。 在用户端,獲取語音數據機1 0 8以絞線對連結1 0 9連接 至用户迴路102。電話設備110典型地均是標準電話,其 透過標準尖環類比語音線路U1連接至獲取語音數據機 108中之語音介面;數據裝置112通常爲電腦,其透過乙 太網路之乙太網路介面連結113連接至獲取語音數據機 1 0 8中之數據介面。爲了便於討論,我們假設每個語音線 路111僅接有一台電話裝置110。POTS電話1 1 4也連接至 用户迴路1 02。 POTS濾波器1 1 5及1 1 6可保護頻率高於語音頻帶之信號 免於被局端交換機1 04與POTS電話114之電話服務干擾。 在局端,獲取語音平台101透過PCM數位中繼器105 , 接收從局端交換機104而來的PCM格式之語音資料,另 透過乙太網路介面連結106,接收從數據網路而來之數據 封包。先以8仟赫茲之網路參考定時信號將該PCM語音資 料予以解碼,接著再加以封_包化,封包的過程包含.,將相 當於分派數位中繼器時槽之定址資訊窝入封包表頭中;随 後,透過雙線連結103與109以及用户迴路102,以數位 調變線路碼將語音封包與數據封包傳送至獲取語音數據機 1 0 8。此外,還下行傳送一個與該網路定時參考信號同步 之定時參考信號至獲取語音數據機108 a -13- 未紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公笼) (請先閱讀背面之注意事項再填寫本頁) ! i I 訂,! I -----, 466864 Λ7 B7 經濟部智慧財產局員工消費合作社印製 五、發明說明(11 ) 該用來透過用户迴路102,傳送語音封包與乙太網路數 據封包於獲取平台10〗與獲取語音數據機1〇8間之特定數 位調變線路碼,與本發明的是相同的。具下列能力的數位 調變線路碼都是可以使用的:在同—個實體用户迴路上, 其載波頻帶可與其他服務(像是,POTS )的頻帶相隔離; 有足夠的頻寬支援所希望擁有的電話設備與數據裝置數; 功率頻譜密度在聯邦通訊委員會的规範與調節範圍之内; 以及有能力傳輸網路定時參考信號。 爲了確保語音的易了解性,傳送在用户迴路102上時, 獲取語音平台101賦予語音封包較數據封包爲高的優先順 序。此目的是可以達到的,只要,譬如,不斷地傳送等待 中之語音封包,而僅於沒有與語音封包在等待傳送時,才 傳送數據封包。 獲取語音數據機108將透過雙線連結109而來之語音與 數據封包予以接收。數據封包透過乙太網路連結1 1 3,傳 送至數據裝置112。語音封包則是先被打破回pcM語音 樣本的型態,然後再以該定時參考信號作爲時脈,將 P C Μ語音樣本轉換成語音頻帶類比訊號。透過連結 111 ’該語音頻帶類比訊號被傳送至該已定址之電話設備 110° 上行方向則是,獲取語音數據機108中的語音介面電路 會接收從電話設備110中,透過雙線連結111而來之標準 尖環類比訊號。然後,再根據該網路定時參考信號,以8 仟赫茲的速率將該標準尖環類比訊號予以取樣並PCM編 ™ 14 - 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公楚〉 I ^ --------^---------線. t請先閱讀背面之注意事項再填寫本頁) Α7 4 β β 8 G 4 _______Β7___ 五、發明說明(12 ) 碼。這些PCM樣本會被封包化,並透過雙線連結1〇9與 1 0 2以及用户迴路1 〇 2 ’以數位調變線路碼將p c Μ封包傳 送至獲取語音平台101。在此同時,獲取語音數據機1〇8 中的乙太網路介面電路會接收從數據裝置112中,透過連 結1 1 3而來之乙太網路數據封包。如果有必要,這些乙太 網路數據封包會被分段,然後才再以數位調變線路碼,透 過用户迴路102,傳送至獲取語音平台1〇1。 與獲取語音平台101相同的,獲取語音數據機1〇8在透 過用户迴路102傳送封包時,其所使用的優先順序方法, 也是賦予語音包較高的傳送優先順序。 獲取語音平台1 0 1在透過雙線連結1 0 3,接收到該P c Μ 樣本格式之語音封包後,會將之解封包並以該網路定時參 考信號爲時脈,透過數位中繼器1〇 5傳送出去。映射表將 用户端數據機108上之電話線與數位中繼介面時槽予 以對應。彳/t...數據裝置112而來之數據封包則會被重组(如 果有必要),然後並透過乙太網路連結106,傳送至數據 網路。 經濟部智慧財產局員工消費合作社印製 在典型的實施中’還會有一或多個的POTS電話1 1 4與本 發明共享用户迴路102。·該等POTS電話會傳送標準的語 音頻帶類比訊號,至局端交換機104。POTS濾波器1 1 5與 1 1 6均爲低通濾波器,其可避免本發明之較高頻率信號進 入局端交換機104以及POTS電話1 1 4。 圖2是本較佳具體實施例中獲取語音平台1 〇 1之一般性 方塊圖。語骨處理器卡204接收透過四- DS1數位中繼連 -15- 本紙張尺度適用中國國家標準(CNS)A4規格(210 x 297公釐) 4 6 6 86 4 Λ7 B7 經濟部智慧財產局員工消費合作社印製 五、發明說明(13 ) 結1 Ο 5而來的p c Μ格式語音資料。該P c Μ格式語音資料 會被轉換成語音封包,稍後這些封包就會被封裝成乙太網 路封包,而這些乙太網路語音封包就會透過乙太網,路介面 串列連結2 0 5而被傳送至數據處理器卡2 〇 1。乙太網路表 頭及語音封包表頭中之定址資訊,會指引該封包走至相關 於該DS1時槽之電話設備110。上行方向則是,語音處理 器卡2〇4會接收該透過連結2〇5,從數據處理器卡2〇1而 來之乙太網路語音封包’予以緩衝暫存,並以PCM格式 將之平移至DS1中繼器105上的適當時槽中。 語音處理器卡204提供多種不同的功能。它抽取出公眾 交換電話網路之網路定時參考信號,透過連結2 〇 3將之分 配給線路卡200。它提供出時槽的交換以及DS1介面電 路’該介面電路透過DS1中繼器1〇5,將PCM格式之語 骨資料叉換於局端交換機104。語音處理器卡2〇4上具有 主中央處理器,該中央處理器將該動態或靜態的中繼器 DS0提供给線路映射表,以及爲數據處理卡2〇1上之乙太 網路又換器提供乙太網路交換管理。主中央處理器還執行 基本的管理介面功能,像是爲對向處理器、數位信號處理 器以及可程式邏輯装置,執行軟體負擔映像的分配。卩匸工 匯流排2 0 6負貴將主中央處理器的控制與命令訊息運送至 數據處理器卡201。 數據處理器卡201主要是充作乙太網路交換器,其爲乙 太網路數據封包,在乙太網路介面連結1〇6之數據網路與 連結2 0 2之適當線路卡200間選擇傳送路徑;其也爲乙太 {請先閱讀背面之;tii意事項再填寫本頁} _ I I ! I I I I I I ,1 I — It . -16-I -5- This paper size is in accordance with China National Standard (CNS) A4 (210 X 297 mm) (Please read the legal notices on the back before filling out this page) Loading * ----- 1— Order ·- -· Line 466864 A7 V. Description of the invention (3) and phase modulation and quadrature amplitude phase modulation. The problem with this method is the digital / analog / digital conversion performed by V at the central office from the blog code modulation (pcM) digital signal to an analog loop signal, and then from the analog loop signal back to the DAML digital signal. It will cause quantization error and phase misalignment, which will lead to the degradation of signal quality. Another technology that digitally transmits multiple voice lines over a subscriber loop is the Integrated Services Digital Network (ISDN). It is a system of multiple voice / data channels with virtual digits and signal channels. However, 1301 ^ needs to change the switching system equipment ’and the management and maintenance method of the switching system. Another method is to send voice packets over a data network that contains user circuits. This method is better known as transmitting voice over Internet Protocol (VOIP), transmitting voice over asynchronous transmission mode (VOATM), and transmitting voice over frame transfer (VOFR). VOIP—usually used in the campus environment. The wire used in each terminal is the EIA / TIA-570-91 standard published by the Electronics Industry Alliance / Telecommunications Industry Association in June 1991: "Home and Light Commercial Electricity "jj wire" type 5 wire or optical fiber; each terminal is connected to a universal switching structure 'like Ethernet, asynchronous transmission mode or hybrid system. In addition, through a gateway such as a router or a layer 3 switching system, the campus network can be called to bridge to the Internet or the corporate network. In some cases, desktop computers and other devices can be used as VOIP enabled terminals. We use enabled terminals to support compliance with the ITU-T standard published by the International Telecommunication Union (ITU) in February 1998. H.323 " 封包 塑 -6- This paper has the standard Chinese National Standard (CNS) A4 (21〇x 2g7 mm). Please read the precautions on the back of Jing before filling this page.) -------- --- Order ---- ^ ----- line · $ Printed by the Consumer Cooperatives of the Ministry of Economics and Intellectual Property Bureau Printed by the Consumer Cooperatives of the Ministry of Economics and Intellectual Property Bureau Printed by 466864 A7 2. Description of the invention (4) Remote communication of multimedia communication system. Such systems typically use digital signal processors (DSPs) on desktop computers to perform voice "packet compression," and then feed them to other workstations on the corporate intranet in the area or send them to remote locations via the Internet. V〇atm and VOFR are another decapsulation technology that uses the public switched telephone network (pSTN) to transmit voice and network connection. The user circuit can use the signal and transmission technology of the digital user circuit series to extend to the wide area network For V0IP applications. The technology of the digital user circuit series, the bandwidth sent to the client can co-exist with the baseband 卩 〇 for services. ISDN can also provide bandwidth to homes connected to the packetized network. Packetized network! ^! ^ Can also provide voice services. ISDN or digital user circuit technology (such as asymmetric digital user circuits and European high-speed digital user circuits) can also transmit 1? Packets on the user circuit, ATM cells or other frame formats. However, the basic requirements for voice and data are different for this web service. Voice transmission only needs to be small Bandwidth, but the supplied bandwidth must basically be dedicated or continuous with very little delay, delay variation, or loss. Even if the delay is in the range of milliseconds, it will produce a clear echo in the conversation. Intermittent. For example, delays caused by routers and gateways will adversely affect voice. Packetized voice belongs to the category of real-time traffic data, so the basic requirements for transmission of loss and error are very strict. In packet The average end-to-end transmission time of the network must be small, and the change of the end-to-end transmission time, including the loss problem, must also be small. -7- This paper standard is applicable to the Chinese National Standard (CNS ) A4 size (210 X 297 mm) II A 'Binding --------- line · (Please read the legal notice on the back before filling this page) 4 6 6 86 4' Μ 'Ministry of Economy Printed by A7 B7, Consumer Cooperatives of the Property Bureau. 5. Description of the invention (5) In the voice transmission, the overall delay should not exceed 2 (0) milliseconds, which is the acceptable number for commercialization. 100-200 milliseconds is general The goal pursued If the delay is around 800 milliseconds, it will prevent normal conversation. The normal state sinks down. For the delay of 200-800 milliseconds, if it occurs infrequently and occurs during a short call, it can be conditionally accepted. In the traditional voice network, the round-trip delay time is about 20-30 milliseconds. The voice delay in the frame transfer network can be about 125-20,0 milliseconds. The Ethernet network carries TCP / IP The delay time in the packet depends on the volume of the traffic. The range of its change is wider. Because the shared data network technology lacks the timeliness of the network, the above-mentioned delay requirements must be extended to users. It is a severe challenge to transmit voice in the general campus network environment. In addition, ATM as a standard still lacks support for voice compression, silence suppression, suppression of idle channel cells, and lack of support for signals that include converting voice signals to exchange virtual link ATM signals. In addition, because the AT delay network and the cell delay variation introduced by the cell combination / unpacking delay will be accumulated by the ATM repeater buffer used for narrowband services (such as voice services), except for the ATM network itself In addition to the resulting delay, the ATM repeater also introduces some additional delay. It is therefore an object of the present invention to provide a system that can provide additional voice and data lines on a user circuit shared with POTS. To this end, it is another object of the present invention to provide support for any in-band signal that supports advanced telephone characteristics. -8- This paper size is in accordance with Chinese National Standard (CNS) A4 (210 X 297 mm) — — — — — 1 — --- — II Order -----] I JN ^ (Please read the Please fill in this page again for attention> 4 6 6 864 A7 __ B7 V. Description of the invention (6) It is another object of the present invention to provide a reconstructed voice signal with almost no attenuation for this system. Provide voice traffic to this system Priority over data traffic is another object of the present invention. To this end, the system provides a line analog modem transmission that does not exceed the 533,300-bit-per-second limit required by the Federal Communications Commission for analog lines. Speed is another object of the present invention. It is another object of the present invention to provide the system with a feature that can avoid signal damage due to additional digital / analog conversion. This system provides a lower data network environment The use cost of common components is another object of the present invention. To provide the system with the feature that the user can install it at the user end is another object of the present invention. Outside voice services It can also be implemented gradually in the network without the need to build an independent data transmission underlying structure in addition to the existing central computer room switch. Summary of the invention The invention is a system that can be used in a single two-line user loop An additional voice line is provided on the 'JL. At the same time, the POTS service of the user circuit can still be retained on this circuit. This system includes a digital modem in the central computer room (c)' which communicates with the user terminal through the user circuit. The other-digital modem connection. The central office (central computer room) digital modem is connected to the central office switch through a direct digital interface, and then connected to the public switched telephone network through the switch. This connection method removes most of the previous techniques. The amount of paper that can be seen in the system-9- The size of the stupid paper is applicable to the Chinese National Standard (CMS) A4 ^ each (210 X 297 public) (please read the precautions on the back before filling this page) --- Order ------ 1-line. Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs 466864 A7 B7 V. Description of Invention (7) Digital / analog conversions other than these. These digital / analog conversions are signal attenuation The primary cause of weakness is' so even if only one such conversion is removed, it will provide the system with a better signal than the previous technology system. The local modem can also use a decapsulation interface (such as Ethernet ) Connect to the data network. The central office modem receives the voice data from the central office switch through the digital relay interface. The voice data is in PCM format, and the network timing reference signal is Clock. Then, the PCM voice sample will be sub-encapsulated by the central office modem and transmitted to the client terminal using the appropriate digital modulation line code. There is a table in the central office modem which records the digital Correspondence between the trunk interface time slot and the telephone line address of the client modem. The central office modem will also synchronize a timing reference signal with the network timing reference signal? Transfer to the client modem. In the upstream direction, the PCM sample voice packet from the client modem is received first, and then it is unpacked and provided to the central office switch through the digital relay interface. At this time, the voice is in PCM format and networked. The timing reference signal is its clock. At the client, one or more telecommunications devices are connected to the voice interface in the digital modem. These telecommunications devices may include telephone equipment, such as traditional telephones, " smartphones ", analog modems, or duplicators. The PCM sample voice packet sent from the central modem will be received by the client modem, and then the timing reference signal will be used as the clock to convert the voice packet into an analog voice band signal; it will then be transmitted to the Addressed telephone equipment. In the upstream direction, the timing reference signal is used as the clock, and the analog voice band signal from the telephone device is converted into -10-. This paper size applies to the Chinese National Standard (CNS) A4 specification (210 X 297 g). (Please (Please read the precautions on the back before filling this page). -------- Order. Printed by the Intellectual Property Bureau of the Ministry of Economic Affairs. Consumer Cooperatives. Printed 466864 A7 B7 Employees of the Intellectual Property Bureau of the Ministry of Economic Affairs. Printed by the Consumer Cooperative. 5. Description of the invention (8) PC Μ samples; then PCM samples are sub-packaged into voice packets; then the client modem uses digital modulation line code to transmit the voice packets to the central office data through the user loop. machine. On the client side, the data device can be connected to the client modem through the data interface. The data device can be any source of data packets, such as a computer, a bridge, a router, or a hub followed by many independent computers. The data packets received by the client modem from the central modem will be sent to the addressed data device via routing. If it is in the uplink direction, the digital modulation line code will be used to transmit the data packet generated by the data device to the central office modem through the user loop. The only non-digital transmission part in this system architecture is the user. An analog loop connecting a telecommunications device to a customer modem. But the distance of this route is usually only a few feet. In the traditional system, the signal loss and damage due to the existence of an analog path between the telephone equipment and the central office switch are greatly reduced due to the digital nature of the architecture. In the present invention, the 'voice and data packets are transmitted to the central computer room through the client loop. However, the main purpose of the present invention is to preserve the intelligibility of youth communication. The main purpose of distinguishing the voice and data in the local and client modems through the voice and data interface and the use of the packet transmission priority method which gives voice packets a higher priority than the data packets is achieved. U.S. Patent No. 5,692,035 t〇〇, Mahoney et al 'and Campbell, Couthorne, Hutchinson, Computer Communications Review, April 1994, Volume 2, 24 " Quality of Service Architecture " There are examples describing such a packet-type system that can give a certain packet priority to a certain transmission priority. This cover -11---- "------------- {Please read the notes on the back before filling this page) Order: Line. This paper size applies to China National Standard (CNS) A4 specifications ( 210 297 mm) 466864 A7 B7 V. Description of the invention (9 dagger transmission priority method, in the presence of data, ensure that the transmission time of speech is still low enough, thus ensuring easy to understand speech, in addition, This method also overcomes the problem of easy speech comprehension encountered in V0IP, VOATM, and V0FR. Description of the Figures Figure 1 is a block diagram of a preferred embodiment of the present invention. Figure 2 is a block diagram of a voice platform of the present invention. Fig. 3 is a block diagram of a voice processing card in the voice acquisition platform of the present invention. Fig. 4 is a block diagram of a data processing card in the voice acquisition platform of the present invention. Fig. 5 is a circuit in the voice acquisition platform of the present invention. Block diagram of the card. Figure 6 is a block diagram of the voice data acquisition machine of the present invention. Figure 7 is a second specific embodiment of a block diagram of the voice processing card in the voice acquisition platform of the present invention. Block diagram of data processing card in the platform The second specific embodiment of the invention. Description of the preferred embodiment Figure 1 is a block diagram of the preferred embodiment of the present invention. For convenience, the central computer room in this figure is only connected to a single user. Generally, multiple users are actually connected At the central office, the voice platform 101 is connected to the user circuit 102 with a twisted pair connection 103, and the voice platform ιοί is connected to the central office switch 104 with a direct digital pcM relay connection. This relay The connection uses the North American standard 8-bit uncompressed shrinking pc Μ technology; In addition, -12- This paper size applies to China National Standard (CNS) A4 specifications (210 X 297 cm) (Please read the back Please fill in this page again for attention) Packing -------- Order ----- Printed by the Consumers 'Cooperative of the Intellectual Property Bureau of the Ministry of Economic Affairs 4 6 6 8 6 4 Printed by the Consumers' Cooperative of the Intellectual Property Bureau of the Ministry of Economic Affairs A7 B7 V. Description of the invention (10) The voice platform 101 is also connected to the data network by link 106. The central office switch 104 is connected to the user circuit 102, which is also connected to the public switched telephone network through the relay link 107. On the client side, get the voice The modem 108 is connected to the subscriber loop 102 with a twisted pair connection 109. The telephone equipment 110 is typically a standard telephone, which is connected to the voice interface in the voice modem 108 through a standard pointed ring analog voice line U1; The data device 112 is usually a computer, which is connected to the data interface in the voice modem 108 via the Ethernet interface link 113 of the Ethernet. For the sake of discussion, we assume that each voice line 111 is connected to only one The telephone device 110. The POTS telephone 1 1 4 is also connected to the subscriber circuit 102. The POTS filters 1 1 5 and 1 1 6 can protect signals with frequencies higher than the voice frequency band from being interfered by the telephone service of the central office switch 104 and the POTS telephone 114. At the central office, the voice platform 101 receives voice data in PCM format from the central office switch 104 through the PCM digital repeater 105, and receives data from the data network through the Ethernet interface link 106. Packet. The PCM voice data is decoded with a network reference timing signal of 8 Hz, and then packetized. The packetization process includes. The addressing information equivalent to the time slot of the digital repeater is assigned to the packet table. Then, through the two-wire connection 103 and 109 and the user circuit 102, the voice packet and the data packet are transmitted to the voice data acquisition device 108 by digitally modulating the line code. In addition, a timing reference signal synchronized with the timing reference signal of the network is also transmitted to the voice data acquisition unit 108 a -13- The paper size is applicable to China National Standard (CNS) A4 (210 X 297 male cage) (please first Read the notes on the back and fill out this page)! I I order! I -----, 466864 Λ7 B7 Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs 5. Description of the invention (11) This is used to transmit voice packets and Ethernet data packets to the acquisition platform 10 through the user circuit 102. It is the same as that of the present invention for acquiring a specific digital modulation line code between the voice data modem 108. Digital modulation line codes with the following capabilities are available: on the same physical user circuit, the carrier frequency band can be isolated from the frequency bands of other services (such as POTS); there is sufficient bandwidth to support the desired The number of telephone equipment and data equipment owned; the power spectral density is within the specifications and adjustments of the Federal Communications Commission; and the ability to transmit network timing reference signals. In order to ensure the easy understanding of the voice, when transmitting on the user circuit 102, the acquiring voice platform 101 gives the voice packet a higher priority than the data packet. This purpose can be achieved, as long as, for example, voice packets are waiting to be transmitted continuously, and data packets are transmitted only when there are no voice packets waiting to be transmitted. The voice data acquisition unit 108 receives voice and data packets from the two-line connection 109. The data packet is transmitted to the data device 112 through the Ethernet link 1 1 3. The voice packet is first broken back to the pcM voice sample type, and then the timing reference signal is used as the clock to convert the PC voice sample into a voice band analog signal. Via link 111 ', the analog signal of the voice band is transmitted to the addressed telephone device 110 °. In the upward direction, the voice interface circuit in the voice modem 108 will be received from the telephone device 110 through the two-line connection 111. The standard sharp ring analog signal. Then, according to the network timing reference signal, the standard sharp ring analog signal is sampled and PCM coded at a rate of 8 仟 Hertz. 14-This paper size is in accordance with China National Standard (CNS) A4 (210 X 297 cm) 〉 I ^ -------- ^ --------- line. TPlease read the notes on the back before filling this page) Α7 4 β β 8 G 4 _______ Β7 ___ V. Description of the invention (12 ) code. These PCM samples will be packetized, and the p c M packet will be transmitted to the acquisition voice platform 101 by digitally modulating the line code through the two-line connection 109 and 102 and the user circuit 1 02 '. At the same time, the Ethernet interface circuit in the voice data acquisition device 108 receives the Ethernet data packet from the data device 112 through the connection 1 1 3. If necessary, these Ethernet data packets are segmented, and then the line code is digitally modulated, transmitted through the user loop 102, and transmitted to the voice acquisition platform 101. Similar to the acquisition of the voice platform 101, when the acquisition voice data machine 108 transmits packets through the user circuit 102, the priority method used by the acquisition also gives the voice packets a higher transmission priority. Get the voice platform 1 0 1 After receiving the voice packet in the P c M sample format through the dual-line connection 103, it will unpack it and use the network timing reference signal as the clock to pass through the digital repeater 105 sent out. The mapping table maps the telephone line on the client modem 108 to the time slot of the digital trunk interface.彳 / t ... Data packets from the data device 112 are reassembled (if necessary), and then transmitted to the data network via the Ethernet link 106. Printed by the Consumer Cooperative of the Intellectual Property Bureau of the Ministry of Economic Affairs. In a typical implementation, there will also be one or more POTS phones 1 1 4 sharing the user circuit 102 with the present invention. · These POTS phones will send standard audio and video analog signals to the central office switch 104. The POTS filters 1 1 5 and 1 1 6 are low-pass filters, which can prevent higher frequency signals of the present invention from entering the central office switch 104 and the POTS telephone 1 1 4. FIG. 2 is a general block diagram of the voice platform 101 obtained in the preferred embodiment. Bone processor card 204 receives through four-DS1 digital relay connection -15-This paper size applies Chinese National Standard (CNS) A4 specifications (210 x 297 mm) 4 6 6 86 4 Λ7 B7 Staff of Intellectual Property Bureau, Ministry of Economic Affairs Printed by the Consumer Cooperative V. Invention Description (13) Concluded in the PC M format voice data from 105. The P c Μ format voice data will be converted into voice packets, and later these packets will be encapsulated into Ethernet packets, and these Ethernet voice packets will be serially connected through the Ethernet interface. 2 0 5 is transferred to the data processor card 2 01. The addressing information in the Ethernet header and the voice packet header will direct the packet to the telephone device 110 related to the DS1 time slot. In the upward direction, the voice processor card 204 will receive the Ethernet voice packet 'from the data processor card 201 through the link 2050' to buffer temporarily and store it in PCM format. Pan to the appropriate time slot on the DS1 repeater 105. The voice processor card 204 provides a variety of different functions. It extracts the network timing reference signal of the public switched telephone network and distributes it to the line card 200 through the link 203. It provides time slot exchange and DS1 interface circuit. This interface circuit crosses the PCM format bone data to the central office switch 104 through the DS1 repeater 105. The voice processor card 204 has a main central processing unit, which provides the dynamic or static repeater DS0 to the line mapping table, and changes the Ethernet on the data processing card 201. The device provides Ethernet exchange management. The main CPU also performs basic management interface functions, such as the allocation of software burden images to the on-board processor, digital signal processor, and programmable logic device. The handy bus 2 0 6 transports the control and command messages of the main CPU to the data processor card 201. The data processor card 201 is mainly used as an Ethernet switch, which is an Ethernet data packet. The Ethernet interface is connected to a data network of 106 and an appropriate line card of 200. Select the transmission path; it is also the Ether (please read the back first; fill in this page with the relevant information) _ II! IIIIII, 1 I — It. -16-
4 66 864 A7 五、發明說明(14 ) 網路語音封包,在乙太網路 ^^丨_運結205心語音處理器卡 2〇4與連結202之適當線路卡⑽間選擇傳送路徑。 線路卡200負貴將絞線對連結1Q3集中在中央機房裏的 王架上,連至用户迴路102。透過多重的尖環連結1〇3, 每一個線路卡200典型地均支援多個獲取語音數據機 108。爲万便説明,圖中所示之線路卡2GG每—個均透過 八尖環連結103,支援八個獲取語音數據機1〇8。 在透過用户迴路102,於獲取語音平台1〇1與獲取語音 數據機108間傳送資料時,線路卡2〇〇負貴的是,局端侧 的數位線路碼的處理;另外,它也負貴處理語音封包以及 乙太網路數據封包。乙太網路數據封包與乙太網路語音封 包乃疋透過全雙工乙太網路介面乙太網路後平面連結 202,從數據處理器卡201處接收而來的。我們使用該數 位線路碼,將乙太網路數據封包透過連結1〇3,傳送於適 當的用户迴路102上β將乙太網路語音封包的乙太網路封 套去除之後’也可以使用該數位線路碼,透過連結〗〇 3, 傳送於適當的用户迴路1〇2上。, 在上行方向中’線路卡2 0 0透過連結1 〇 3,會接收到與 它相關的獲取語音數據機108所傳送過來之以數位線路碼 格式存在的語音封包與乙太網路數據封包。線路卡2〇〇先 將該線路碼解碼,將該語音封包封裝成乙太網路封包,然 後,再透過乙太網路後平面連結202將該乙太網路語音封 包與乙太網路數據封包傳送至數據處理器卡2〇1。 -17 本紙張尺度適用中國國家標準(CNS)A4規格(210 χ 297公釐〉 閲 讀 背 £ 之 注 項 再 I裝 頁 訂 經 濟 部 智 慧 財 產 局 員 工 消 費 合 作 社 印 製 4 6 6 864 Α7 ------- . Β7 五、發明說明(15 ) 從語音卡204送至連結203上之網路定時參考信號 8K—NTR,則是作爲線路卡2 0 0之時脈信號。. 圖3是語音處理器卡204之方塊圖。在操作時,線路介 面303透過了數位中繼器1〇5,接收到pcjyj語音資料,·然 後透過串列連結3 0 5,將之傳送至數位信號處理器3 〇 j。 線路介面303是由一四- DS1線路框架器,一可支援四個 每办1.544百萬位元資料串流之線路介面部件以及一磁性 部件所構成的。透過本地處理器匯流排3 η與p c j匯流排 206 ’主中央處理器3〇4掌控著該等線路介面3〇3部件。 數位信號處理器3 01是該PCM語音資料的緩衝器,並在 DS1時槽基座上將其轉換爲語音封包,然後再將每一個語 音封包封裝成乙太網路封套型式。數位信號處理器3〇1使 用靜態隨機存取記憶體SRAMs 308來儲存程式碼與資料》 數位信號處理器3 0 1還具有回音消除功能。主中央處理器 304亦負貴數位信號處理器301映射表的管理,此表格記 載著DS1時槽與線路卡200媒體存取控制(MAC )位址/獲 取語音線路識別碼的對應關係。然後,該乙太網路語音封 包就會透過資料匯流排連結302而被傳送至乙太網路控制 器300,以便進一步地再透過乙太網路介面串列連結 205,傳送至數據處理器卡201。20百萬赫振盪器309則 會將系統的時脈信號提供給乙太網路控制器3 0 0。 在上行方向中,透過連結20 5,乙太網路控制器300會 接收到由數據處理器卡2〇1而來之乙太網路語音封包。透 過資料匯流排連結3 〇 2,該乙太網路語青封包會被傳送至 -18- 本紙張尺度適用t國國家標準(CNS)A4規格<210 297公釐〉 (請先閱讀背面之沒意事項再填寫本頁) 裝--------订-- -----線- 經濟部智慧財產局員工消費合作社印製 4 B 6 86 4 Λ7 B7 五、發明說明(16) 數位信號處理器30卜數位信號處理器3〇1會將該語音封 包(乙太網路封套予以去除,並將去除封套後之語音封包 解封,再將解封出來之PCM語音資料予以‘緩衝暫存,隨 後再將該PCM資料,透過線路介面3〇3傳送至數位中繼 器105之DS1上。數位信號處理器3〇1接受主中央處理器 3〇4的命令,提供DS1語音時槽交換功能。 在此較佳具體實施例中,每—個乙太網路控制器3 〇 〇所 採用的均是賽若斯邏輯公司之型號爲CS89〇〇的乙太網路 控制器,每一個數位信號處理器3〇1所採用的均是德州儀 器公司之型號爲TMS320C6201的數位信號處理器。每一個 數位信號處理器301均乃透過其上之兩個多通道緩衝串列 埠(McBSPs )以及串列連結3 0 5,而連接至D s丨線路介面 303。在此較佳具體實施中,每一個數位信號處理器3〇1 上的兩McBSPs均與中繼器105之四DS1中的某一個有 關°市場上供應許多種類之分立與集成的線路介面。 在此較佳具體實施中’數位中繼器1〇5是一個使用GR_ 303信號協定之四-DS1介面,該信號協定描述於泰爾‘科 迪拉科技公司1998年十二月,議題2,|’集成數位迴路載 波系統的一般要件、目標與介面"之標準刊物GR-303中。 雖然,本具體實施例所採用的中繼器1 〇 5至局端交換機的 傳送協定是,GR-303 DS1中繼連結,但是任何合適的可 直接數位地連接至該交換構體之協定與實體連結,均是可 以使用的。譬如,泰爾科迪拉科技公司1994年十月,公告 1,版次1,議題2," SLC-96®數位迴路載波系統與本地數 -19- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) (請先閲讀背面之注意事項再填寫本頁) 裝--------訂-----I---線. 經濟部智慧財產局員工消費合作社印製 4 6 6 8 6 4 A7 __B7 經濟部智慈財產局員工消費合作社印製 五、發明說明(17) 位交換機間之數位介面"之標準刊物TR-TSY-000008中所 描述的TR-008協定’也是可以使用〇 主中央處理器304透過現用與備用鑲嵌式操作通道 (EOCs)以及時槽管理通道(TMCs),來管理該GR-303協 定。負貴中繼器105中DS1-1通信之連結307a,與連結 305a ’爲現用的EOC及TMC通道,而負貴中繼器1〇5中 DS卜2通信之連結307b,與連結305b,則爲備用EOC及 T M C通道。在此較佳具體實施例中,每一個連結3 〇 7均 連接到主3 0 4上的T D Μ埠,我們可以用程式來抽取該 EOC及TMC訊息通道並將其傳遞至SCC,在此處的訊息 均會被抽取出來,供應至該主中央處理器3〇4。然後,透 過PCI匯流排2 0 6與本地處理器匯流排3 1 1之傳遞訊息, 將該線路卡200定址與語音時槽交換映射表予以設定,主 中央處理器3 0 4可做出適合的D S 1時槽對應關係。 主中央處理器304還可以針對獲取語音平台1〇1,執行 主中央處理器元件管理功能。習於此藝人士須了解這些功 能包括,下命令,資料的蒐集以及款體的下載。一個連接 至該主中央處理器的本地技藝介面(未顯示),通常也是本 系統的一部分。在本較佳具體實施例中,主中央處理器 3 04是摩托羅拉MPC860ElSr »元件管理訊息位於頻帶内, 其透過PCI匯流排206傳遞給數據處理卡201,透過PCI 匯流排2 0 6與本地處理器匯流排3 1 1傳遞給語音處理卡 204,透過後平面202以乙太網路封包型式傳遞給線路卡 200。PCI橋接器310讓主中央處理器304透過本地處理 -20- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公爱) <猜先閱讀背面之注意事項再填寫本頁) 裝 ----訂---------線‘ -n —1 n ϋ- -n n n Λ 6 6 86 4 Λ7 Β7 五、發明說明(18 ) 器匯流徘3 n,連接至P C丨E流排2 0 6。可以不需要p C j 橋接器3〗0’此取決於所選元件的性質。從數據處理器卡 2 0 1而來,出現在連結4 0 6上之3 3百萬赫定時信號,爲 P CI橋接器3 1 〇提供出時膝。 就如圖式説明的,線路介面3 0 3.抽取該定時信號,以產 生8仟赫兹之網路定時參考信號8K_NTR以供線路卡2 〇 〇使 用。該原始的定時信號是由線路介面3 〇 3從中繼器丨〇 5之 DS1-1中所抽取出來的,並透過連結3〇5a,傳送至相鎖迴 圈電路3 0 6,此電路會予以除頻而送出一個穩定的8仟赫 兹網路定時參考信號8K_NTR,透過連結203至線路卡 2 〇 0。爲具備有備份能力以防ds 1 ·1失誤現象的發生,我 們還可以將線路介面3 〇 3設定成可從譬如,DS1-3中抽取 出該原始定時信號,並將其傳送至一個软體的2選1多工 器,此多工器除了接收DS1-3而來之原始定時信號外,也 可接收從線路介面303 DS1-1而來之原始定時信號,並輸 出此二者之一至相鎖迴圈3〇6。 圖4是數據處理器卡201之方塊圖。數據處理器卡201 透過乙太網路實體介面402與乙太網路介面連結106,接 收從數據網路而來之乙太網路數據封包。然後,透過連結 403,將該乙太網路數據封包傳送至快速乙太網路交換器 4 0 1。然後,快速乙太網路交換器4 〇 1就會以p CI匯流排 206爲傳送路徑,爲該乙太網路數據封包選徑至乙太網路 交換器400 ;交換器則再以乙太網路介面後平面連結202 -21 - 本,纸張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) .---一----------裝--- (請先間讀背面之注意事項再填寫本頁) * -線: 經濟部智慧財產局員工消費合作社印製 4 6 6 86 4 Δ7 Α7 ------ Β7 經濟部智慧財產局員工消費合作社印製 五、發明說明(19) 爲傳送路徑,將乙太網路數據封包選徑至適當的線路卡 20 0 〇 v 乙太網路交換器401透過乙太網路連結205,接收從語 音處理器卡2 0 4而來之乙太網路語音封包。然後,以連結 2 0 2爲路徑,將該乙太網路語音封包選彳f至適當的線路卡 200 〇 在上行方向中,線路卡200透過後平面連結202,傳送 語音及數據乙太網路封包至乙太網路交換器400。乙太網 路數據封包會透過P C I匯流排2 0 6,被選徑至快速乙太網 路交換器401,快速乙太網路交換器則再透過連結403, 將乙太網路數據封包.傳送至乙太網路實體介面402,以便 它們可透過乙太網路介面連結106,傳送至數據網路。從 線路卡2 0 0處所接收而來之乙太網路語音封包,則會被乙 太網路交換器400以乙太網路連結205爲傳送路徑,選徑 至語音處理器卡204。 乙太網路交換器400與401的乙太網路選徑表,乃是透 過PCI匯流排206,保存在主中央處理器304中。 33百萬赫振盪器404透過連結409,爲PCI匯流排206 電路提供出時序,80百萬赫振盪器405則透過連結410, 爲傳送與接收的定時提供出時序。33百萬赫振盪器404還 透過連結406,爲語音處理器卡204提供出定時信號。25 百萬赫振盪器407則是爲乙太網路實體介面402提供出時 序以作爲它們傳送與接收的時脈° -22- 本纸張尺度適用中國國家標準(CNS>A4規格(210 X 297公釐) (靖先閱讀背面之注意事項再填寫本頁) -S裝4 66 864 A7 V. Description of the invention (14) Network voice packet, choose the transmission path between the Ethernet ^^ 丨 _delivery 205 heart voice processor card 204 and the appropriate line card link 202. The line card 200 collects the twisted pair connection 1Q3 on the king frame in the central computer room, and connects it to the subscriber circuit 102. Through multiple sharp ring links 103, each line card 200 typically supports multiple voice modems 108. For the sake of convenience, each of the 2GG line cards shown in the figure is connected to 103 through an octagonal ring, and supports eight voice modems 108. When transmitting data between the voice platform 101 and the voice modem 108 through the user circuit 102, the line card 2000 is expensive, and the processing of the digital line code at the central office side is also expensive. Handle voice packets and Ethernet data packets. Ethernet data packets and Ethernet voice packets are received from the data processor card 201 through the full-duplex Ethernet interface Ethernet backplane connection 202. We use the digital line code to transmit the Ethernet data packet through the link 103 and transmit it to the appropriate user circuit 102. After removing the Ethernet envelope of the Ethernet voice packet, we can also use the digital The line code is transmitted to the appropriate user circuit 102 via the link [03]. In the uplink direction, the 'line card 2 0 0' will receive the voice packet and Ethernet data packet transmitted in the digital line code format transmitted by the voice data acquisition unit 108 associated with it. The line card 200 first decodes the line code, encapsulates the voice packet into an Ethernet packet, and then transmits the Ethernet voice packet and Ethernet data through the Ethernet backplane connection 202. The packet is transmitted to the data processor card 201. -17 This paper size is in accordance with Chinese National Standard (CNS) A4 specification (210 x 297 mm). Note on the back of the note is printed on page I. Binding is printed by the Intellectual Property Bureau of the Ministry of Economic Affairs and Consumer Cooperatives. 4 6 6 864 Α7 --- ----. Β7 V. Description of the invention (15) The network timing reference signal 8K-NTR sent from the voice card 204 to the link 203 is used as the clock signal of the line card 2000. Figure 3 is the voice Block diagram of the processor card 204. In operation, the line interface 303 receives the pcjyj voice data through the digital repeater 105, and then transmits it to the digital signal processor 3 through a serial connection 3 0 5 〇j. The line interface 303 is composed of a four-DS1 line framer, one line interface component that can support four 1.544 million bit data streams, and one magnetic component. It is connected through a local processor bus 3 η and pcj bus 206 'The main CPU 304 controls these line interface components 303. The digital signal processor 301 is a buffer of the PCM voice data, and it is stored on the DS1 time slot base. Convert to voice packets, and then convert each Each voice packet is encapsulated into an Ethernet jacket type. The digital signal processor 301 uses static random access memory SRAMs 308 to store code and data. The digital signal processor 301 also has an echo cancellation function. The main center The processor 304 is also responsible for the management of the mapping table of the digital signal processor 301. This table records the correspondence between the DS1 time slot and the line card 200 media access control (MAC) address / obtaining the voice line identification code. Then, the B The Ethernet voice packet will be transmitted to the Ethernet controller 300 through the data bus link 302, so as to be further transmitted to the data processor card 201 through the Ethernet interface serial link 205. The 10 MHz oscillator 309 will provide the system clock signal to the Ethernet controller 300. In the uplink direction, the Ethernet controller 300 will receive the data processor card through the connection 20 5 An Ethernet voice packet from 2001. Through the data bus link 3 0, the Ethernet voice packet will be sent to -18- This paper standard is applicable to National Standards (CNS) A4 < 210 2 97 mm> (Please read the unintentional matter on the back before filling out this page) Packing -------- Ordering------ Line-Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs 4 B 6 86 4 Λ7 B7 V. Description of the invention (16) Digital signal processor 30. The digital signal processor 301 will remove the voice packet (Ethernet envelope, remove the envelope after removing the envelope, and then The unsealed PCM voice data is buffered temporarily, and then the PCM data is transmitted to the DS1 of the digital repeater 105 through the line interface 303. The digital signal processor 301 accepts the command of the main CPU 304, and provides the DS1 voice time slot exchange function. In this preferred embodiment, each of the Ethernet controllers 300 uses an Ethernet controller model CS8900 from Cyrus Logic, and each digital signal is processed. The digital signal processor model TMS320C6201 used by Texas Instruments is TMS320C6201. Each digital signal processor 301 is connected to the D s 丨 line interface 303 through two multi-channel buffered serial ports (McBSPs) and serial links 305 thereon. In this preferred embodiment, the two McBSPs on each digital signal processor 301 are related to one of the repeaters 105-4 DS1. Many types of discrete and integrated line interfaces are available on the market. In this preferred embodiment, the 'digital repeater 105' is a four-DS1 interface using the GR_303 signal protocol, which was described in Tel 'Cordilla Technologies, December 1998, Issue 2, | 'General Publication GR-303 for General Requirements, Targets and Interfaces of Integrated Digital Loop Carrier Systems. Although the transmission protocol of the repeater 105 to the central office switch used in this embodiment is a GR-303 DS1 relay connection, any suitable protocol and entity that can be directly digitally connected to the switching fabric Links are all available. For example, Telcordila Technology Corporation, October 1994, Announcement 1, Issue 1, Issue 2, " SLC-96® Digital Loop Carrier System and Local Number -19- This paper standard applies to China National Standard (CNS) A4 specification (210 X 297 mm) (Please read the precautions on the back before filling out this page) Packing -------- Order ----- I --- line. Staff Consumption of Intellectual Property Bureau, Ministry of Economic Affairs Printed by the cooperative 4 6 6 8 6 4 A7 __B7 Printed by the Consumers ’Cooperative of the Intellectual Property Bureau of the Ministry of Economic Affairs 5. Description of the invention (17) TR described in the standard publication TR-TSY-000008 of the digital interface between switches The -008 protocol can also use the main CPU 304 to manage the GR-303 protocol through the active and standby mosaic operating channels (EOCs) and time slot management channels (TMCs). The link 307a for DS1-1 communication in the link repeater 105 and link 305a 'are the current EOC and TMC channels, while the link 307b for the DS link 2 in link repeater 105 and link 305b, then For backup EOC and TMC channels. In this preferred embodiment, each link 307 is connected to the TD M port on the main 304. We can use programs to extract the EOC and TMC message channels and pass them to the SCC, here All the messages will be extracted and supplied to the main CPU 304. Then, the message is transmitted through the PCI bus 2 0 6 and the local processor bus 3 1 1, and the line card 200 addressing and voice time slot exchange mapping table is set, and the main CPU 3 0 4 can make a suitable DS 1 time slot correspondence. The main central processing unit 304 may also perform the main central processing unit component management function for the acquired voice platform 101. Those who are familiar with this art must understand these functions, including ordering, collection of information and download of models. A local technology interface (not shown) connected to the main CPU is usually also part of the system. In the preferred embodiment, the main CPU 304 is Motorola MPC860ElSr »The component management information is located in the frequency band, which is transmitted to the data processing card 201 through the PCI bus 206, and through the PCI bus 206 and the local processor The bus 3 1 1 is transmitted to the voice processing card 204 and transmitted to the line card 200 in an Ethernet packet type through the rear plane 202. The PCI bridge 310 allows the main CPU 304 to process locally. -20- This paper size applies the Chinese National Standard (CNS) A4 specification (210 X 297 public love) < Guess to read the precautions on the back before filling this page). ---- Order --------- line '-n —1 n ϋ- -nnn Λ 6 6 86 4 Λ7 Β7 V. Description of the invention (18) The device bus 3 n, connected to the PC 丨 E Streams 2 0 6. The p C j bridge 3 may not be required. This depends on the nature of the selected element. Coming from the data processor card 201, the 33 MHz timing signal appearing on the link 406 provides the timing for the PCI bridge 3 10. As illustrated in the figure, the line interface 3 0 3 extracts this timing signal to generate a network timing reference signal 8K_NTR of 8 Hz for use by the line card 2000. The original timing signal is extracted from the DS1-1 of the repeater 丨 05 by the line interface 3〇3, and transmitted to the phase-locked loop circuit 3 06 through the link 3005a. This circuit will Divide the frequency and send a stable 8Hz network timing reference signal 8K_NTR, which is connected to the line card 2000 through the connection 203. In order to have a backup capability to prevent the occurrence of DS 1 · 1 errors, we can also set the line interface 3 0 3 to be able to extract the original timing signal from, for example, DS1-3 and send it to a software 2 to 1 multiplexer. In addition to receiving the original timing signal from DS1-3, this multiplexer can also receive the original timing signal from the line interface 303 DS1-1 and output one of the two to the phase lock. Loop 30.6. FIG. 4 is a block diagram of the data processor card 201. The data processor card 201 receives the Ethernet data packet from the data network through the Ethernet physical interface 402 and the Ethernet interface connection 106. Then, through the link 403, the Ethernet data packet is transmitted to the fast Ethernet switch 401. Then, the fast Ethernet switch 401 uses p CI bus 206 as a transmission path, and selects a path for the Ethernet data packet to the Ethernet switch 400; the switch then uses Ethernet Network interface rear plane connection 202 -21-this paper size is applicable to China National Standard (CNS) A4 specification (210 X 297 mm). (Please read the precautions on the back before filling out this page) * -Line: Printed by the Consumers 'Cooperatives of the Intellectual Property Bureau of the Ministry of Economy 4 6 6 86 4 Δ7 Α7 ------ Β7 Employees' Cooperatives of the Intellectual Property Bureau of the Ministry of Economy Printed 5. Description of the invention (19) For the transmission path, the Ethernet data packet is routed to the appropriate line card 20 00v Ethernet switch 401 is connected to 205 through the Ethernet link, and receives voice processing from Ethernet card from the device card 2 0 4. Then, using link 202 as the path, select this Ethernet voice packet to the appropriate line card 200. In the uplink direction, the line card 200 transmits voice and data Ethernet through the back plane connection 202. The packet is sent to the Ethernet switch 400. The Ethernet data packet will be routed to the fast Ethernet switch 401 through the PCI bus 206, and the fast Ethernet switch will then send the Ethernet data packet through the link 403. To the Ethernet physical interface 402 so that they can be transmitted to the data network via the Ethernet interface link 106. The Ethernet voice packet received from the line card 200 will be routed to the voice processor card 204 by the Ethernet switch 400 using the Ethernet link 205 as the transmission path. The Ethernet routing tables of the Ethernet switches 400 and 401 are stored in the main CPU 304 through the PCI bus 206. The 33 megahertz oscillator 404 provides timing for the PCI bus 206 circuit through the connection 409, and the 80 megahertz oscillator 405 provides timing for transmission and reception timing through the connection 410. The 33 megahertz oscillator 404 also provides timing signals to the speech processor card 204 through the link 406. The 25 megahertz oscillator 407 provides the timing for the Ethernet physical interface 402 as the clock for their transmission and reception ° -22- This paper size applies to the Chinese national standard (CNS > A4 specification (210 X 297 Mm) (Jing first read the precautions on the back before filling out this page) -S Pack
I! 一SJ. I — II 線 4-6 6 86 4 Λ7 137 五 發明說明(2〇 經濟部智tt財產局員工消費合作杜印製 匯流排仲裁器408使用簡單的旋轉優先順序法則,來控 制pci匯流排使用者對pci匯流排的控制所有權。本發明 之匯流排仲裁器408是一個可裎式邏輯裝置。視所選元件 的不同,可能不需要—個獨立的匯流棑仲裁器4 0 8 » 在本較佳具體實施中,每一個乙太網路交換器4〇〇所採 用的均是加利莱奥科技公司’.型號爲GT-48001A之乙太網 路交換控制器,快速乙太網路交換器4 〇〗所採用的則是加 利萊奥科技公司,型號爲GT-480〇2A之快速乙太網路交換 控制器’另外,每一個乙太網路實體介面4 〇 2所採用的則 哼是等級一通訊公司,型號爲LXT970之雙速快速乙太網 路收發器。 圖5是線路卡2〇〇的方塊圖。乙太網路收發器5〇7透過 乙太網路介面後平面連結2 0 2,接收從數據處理器卡2 〇 1 而來之乙太網路語音及數據封包。隨後,再透過連結 508 ’將該乙太網路語音及數據封包傳送至處理器5〇2。 20百萬赫振盪器509提供一定時信號至乙太網路收發器 5 0 7及處理器5 0 2。動態隨機存取記憶體DRAM 505及内 可定址記憶體CAM 506則分別地供處理器5 0 2儲存程式與 資料、緩衝封包,以及快速奎閱MAC/IP地址之用。其他 合適的記憶體形式也是可以使用。處理器502會將語音封 包的乙太網路封套予以去除,將大的乙太網路數據封包分 段,然後,透過匯流排連結5 0 3,將該語音及乙太網路數 *封包傳送至數位信號處理器501。- % 裝— {請先閱讀背面之注意事項再填寫本頁) 訂. --線 -23- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) 466864 A7 B7 經濟部智慧財產局員工消費合作杜印製 五、發明說明(21 ) 數位信號處理器5 0 1乃透過主處理器介面(η p I)匯流排 5 0 3而連接至處理器5 〇 2。Η Ρ I匯流排5 0 3包.含一位址與 資料匯流排,一給每一數位信號處理器5〇1及各種控制輸 入信號用之中斷連結。場可程式閘陣列FPGA 504則是作 爲處理器502與數位信號處理器5〇1的控制信號介面。雖 然該傳遞於處理器502與數位信號處理器501間之控制信 號是非常標準化的,但是針對時間以及控制輸出信號與所 希望控制輸入信號的組合上,通常在各廠商之間是有差別 的。場可程式閘陣列5 〇 4將控制輸出信號予以格式化,以 符合欲得控制輸入信號的規格需求。習於此藝人士從場可 程式閘陣列更一般化的名字:”膠邏輯,,,就可深刻地瞭 解到它的功能。視所選擇元件的不同,有可能不必使用該 膠邏輯場可程式閘陣列。 數位信號處理器501會將該語音封包及乙太網路數據封 包,轉換成該數位線路碼,然後將該線路碼資料傳送至類 比前端AFEs 500,以便可透過連結i 〇 3將線路碼資料傳送 於用户迴路1〇2上。從語音處理器卡2〇4而來在線2〇3上 足8什赫網路定時參考信號8艮—NT]R,會爲類比前端5〇〇及 數位信號處理器5 〇 1提供出時間標準。 在上行方向中’透過用户迴路連結103,AFEs 500會接 收到線路碼格式之語音及乙太網路數據錡包6 AFEs 500會 先將孩類比線路碼資料,轉換成數位格式串列位元流,然 後再將該位元_流傳送至相關的數位信號處理器5 0 1的緩 衝串列崞。數位信號處理器5 0 1會先將該線路碼解碼,然 -24 本紙張尺度_中圈固家標以 C二 / y Z X υ I z 份 s * <請先閱讀背面之注意事項再填寫本頁)I! I SJ. I — II line 4-6 6 86 4 Λ7 137 Five invention descriptions (20. Intellectual Property Bureau, Ministry of Economic Affairs, Employee Consumption Cooperation, Du Printed Bus Arbiter 408, using simple rotation priority rules to control PCI bus user control over PCI bus ownership. The bus arbiter 408 of the present invention is a portable logic device. Depending on the selected components, an independent bus arbiter 4 0 8 may not be needed. »In this preferred embodiment, each Ethernet switch 400 uses Galileo Technology Co., Ltd .. Ethernet Switch Controller GT-48001A, Fast Ethernet The network switch 4 〇 uses the Galileo technology company, model GT-480〇2A fast Ethernet switch controller. In addition, each Ethernet physical interface 4 002 The one adopted is a dual-speed fast Ethernet transceiver of the class one communication company, model LXT970. Figure 5 is a block diagram of the line card 200. The Ethernet transceiver 507 passes through the Ethernet Interface back plane connects 2 0 2 and receives from the data processor card 2 1 from the Ethernet voice and data packet. Then, the Ethernet voice and data packet is transmitted to the processor 502 through the link 508 '. The 20 MHz oscillator 509 provides a certain time signal to Ethernet transceiver 507 and processor 502. Dynamic random access memory DRAM 505 and internal addressable memory CAM 506 are used by processor 502 to store programs and data, buffer packets, and Quickly read the MAC / IP address. Other suitable memory forms are also available. The processor 502 will remove the Ethernet envelope of the voice packet, segment the large Ethernet data packet, and then, Through the bus link 5 0 3, the voice and Ethernet data * packet is transmitted to the digital signal processor 501.-% 装 — {Please read the precautions on the back before filling this page) Order. --Line-23 -This paper size is in accordance with China National Standard (CNS) A4 (210 X 297 mm) 466864 A7 B7 Printed by the Intellectual Property Bureau of the Ministry of Economic Affairs and Consumer Cooperation Du V. Invention Description (21) Digital Signal Processor 5 0 1 Main processor interface (η p I) Exhaust stream 503 is connected to the processor 5 billion 2. Ρ PI bus 503 package. It contains one address and data bus, one for each digital signal processor 501 and various interrupt links for control input signals. The field programmable gate array FPGA 504 is used as the control signal interface of the processor 502 and the digital signal processor 501. Although the control signals that should be passed between the processor 502 and the digital signal processor 501 are very standardized, the time and the combination of the control output signal and the desired control input signal are usually different between manufacturers. The field programmable gate array 504 formats the control output signal to meet the specifications of the desired control input signal. Those who are familiar with this art can understand the function of the field programmable gate array from the more general name: "glue logic," depending on the selected components, it may not be necessary to use the logic field programmable The digital signal processor 501 converts the voice packet and the Ethernet data packet into the digital line code, and then transmits the line code data to the analog front-end AFEs 500 so that the line can be connected through the link i 〇3 The code data is transmitted on the user circuit 102. The voice signal from the voice processor card 204 is on-line and on the line 203, and the network timing reference signal 8gen-NT] R will be the analog front-end 500 and The digital signal processor 5 〇1 provides the time standard. In the uplink direction, 'through the user loop connection 103, the AFEs 500 will receive the voice and Ethernet data in line code format 6 AFEs 500 will first analogize the line The code data is converted into a digital format serial bit stream, and then the bit stream is transmitted to the relevant digital signal processor's buffer serial 501. The digital signal processor 501 will first send the line Code decoding, then -24 This paper size _ Middle circle solid label with C 2 / y Z X υ I z copies s * < Please read the precautions on the back before filling this page)
4 66 864 經 濟 部 智 慧 財 產 局 員 工 消 費 合 作 社 印 製 A7 B7 五、發明說明(22 ) 後將解碼出來之5吾音封包及分段的乙太網路數據封包,傳 运至處理器502。孩語音封包會被裝封爲具乙太網路架 構,该分段的乙太網路數據封包則會被重新组合。接著, 處理器502會透過連結508,將該處理好之乙太網路語音 封包及乙太網路數據封包,傳送至乙太網路收發器5〇7。 在本發明中,每一個線路卡均有一個M A C層的地址,以 便數據處理器卡201上的交換行爲可變得較容易,另外也 可使得該獲取語音線路映射至p c M中繼器i 〇 5上適當時 槽的映射行爲,變得較容易。然後,乙太網路收發器5〇7 會透過乙太網路介面後平吊連結2〇2,將所有的乙太網路 語音及數據封包傳送至數據處理器卡2〇1。 在本發明中’每一個現用狀態的(離鉤的)獲取語音數據 機電話線路1 1 0均獨佔用户迴路i 〇 2每秒丨9 2仟位元的頻 寬’其中包含了每秒6 4什位元的上串流,每秒6 4仟位元 的下串流’以及給信令與控制用之每秒6 4仟位元頻寬。 獲取語音數據機數據線路丨i 3中傳送串流的位元速率則不 予限制’它可以使用用户迴路中尚未分.配給現用獲取語音 數據機電話線路用的剩餘頻寬,來傳送乙太網路數據封 包。 在本較佳具體實施中,該用來在獲取語音平台101與獲 取語音數據機108間,傳送該語音封包與乙太網路數據封 包之特定數位調變線路碼,其所使用的編碼技術是帕拉代 恩公司所發展.的三玩。三玩是帕拉代恩公司的註册商標。 二玩技術使用的是ISDN封套内的頻譜,但保留最低可至0 -25- 本紙張尺度適用中國國家標準2—^ (請先閲讀背面之注意事項再填寫本頁)4 66 864 Printed by the Consumer Affairs Cooperative of the Intellectual Property Office of the Ministry of Economic Affairs A7 B7 V. After the description of the invention (22), the 5 voice packets and segmented Ethernet data packets will be transmitted to the processor 502. The child voice packet will be packed into an Ethernet structure, and the segmented Ethernet data packets will be reassembled. Then, the processor 502 sends the processed Ethernet voice packet and Ethernet data packet to the Ethernet transceiver 507 through the link 508. In the present invention, each line card has an MAC layer address, so that the switching behavior on the data processor card 201 can be easier, and in addition, the obtained voice line can be mapped to the pc M repeater i. The mapping behavior of the appropriate time slot on 5 becomes easier. Then, the Ethernet transceiver 507 will suspend the connection 202 through the Ethernet interface, and transmit all Ethernet voice and data packets to the data processor card 201. In the present invention, "every active state (off-hook) acquiring voice modem telephone line 1 1 0 has exclusive user circuit i 〇2 per second 丨 9 2 bit bandwidth" which includes 6 4 per second The upper bit stream of 64 bits, the lower bit stream of 64 bits per second, and the 64 bit bits bandwidth per second for signaling and control. Get the voice modem data line 丨 The bit rate of the transmission stream in i 3 is not limited. It can use the remaining bandwidth in the user loop. The remaining bandwidth allocated to the current voice modem phone line is used to transmit Ethernet. Data packets. In this preferred embodiment, the specific digital modulation line code used to transmit the voice packet and the Ethernet data packet between the voice acquisition platform 101 and the voice data acquisition unit 108, the encoding technology used is Three Plays developed by Paradyne. Sanwan is a registered trademark of Paradyne. The second play technology uses the spectrum in the ISDN envelope, but the minimum can be kept to 0 -25- This paper size applies to Chinese national standard 2— ^ (Please read the precautions on the back before filling this page)
Λ7 4 6 β 8 6 4 ____Η7__ 五、發明說明(23 ) 赫茲之較低頻率,專鬥給P〇TS之傅送使用。三玩產生一 定時參考信號,並透過用户迴路1 0 2,將之信號處理器之 間。該三玩定時參考信號的相位則是鎖定於該透過連結 2 0 3,傳送至數位信號信號處理器之間。該三玩定時參考 信號的相位則是鎖定於該透過連結2 0 3,傳送至數位信號 傳送於獲取語音平台10〗中數位信辦處理器501與獲取語 音數據機1頻率.,專門给POTS之傳送使用。三玩產生一定 時參考信號並透過用户迴路102,將之108中相關的數位 信號處理器之間。該三玩定時參考信號的相位則是鎖定於 該透過連結203,傳送至數位信號處理器501之網路定時 參考信號8K_NTR。在本發明之圖示説明中,該三玩定時 參考信號與網路定時參考信號8K_NTR沒有差別,而圖中 所示之網路定·時參考信號SK_NTR,從獲取語音平台1〇1 到獲取語音數據機1 〇 8 '間是屬非中斷型式的。. 在、本較佳具體實施例中,乙太網路收發器5 〇 7所採用的 是’摩托羅拉MC68160加強型乙太網路串列收發器;處理 齒50 2所採用的是,摩托羅拉之型號爲1^>〇860£1^功率 QUICC之微處理器;每一個數位信號處理器5〇1户斤採用的 則是’德州儀器之型號爲TMS320VC549之數位信號處理 器’加入帕拉代恩公司所提供的軟體,此數位信號處理器 可支极遠二玩線路碼的編碼法則;每一個AFE 5 0 0所採用 的則是,伯布朗型號爲AFE1137,微調成可支援該三玩線 路碼編碼法則之類比前端。 -26 - 本紙張尺度適用中國國家標準(CNS)A4規格(210 x 297公釐) (请先閱讀背面之法意事項存填寫本頁) - -------- 訂---------線. 經濟部智慧財產局員工消費合作社印製 ^6 6 864 A7 ---------B7___ 五、發明說明(24 ) 圖6是本發明之獲取語音數據機1〇8的方.塊圖。電話設 備1 1 0透過雙線連結u〗,連接至用户線路介面電路 (SLICs) 600。用户線路介面電路6〇0將該從電話11()而來 之尖環類比訊號予以接收,取樣該類比訊號,然後使用 P C Μ技術將該信號予以數位式地編碼,最後透過連結 602,將該等語音樣本傳送至處理器6〇ι。取樣頻率是以 8什赫網路定時參考信號8K-NTR爲標準的,該網路定時 參考信號8K_NTR則是數位信號處理器6 〇 4透過連結6 〇 3 傳遞至用户線路介面電路6 〇 〇。取樣資訊則以每秒6 4仟位 元串列位元流(包含8仟個樣本,每樣本具8個位元)的形 式’從用户線路介面電路6〇〇中,透過連結602傳送至處 理器601上之全雙工串列通信埤道介面(;§(:(;^)。 處理器60 1將該等PCM語音樣本分封化爲語音封包,封 包中包含信號資訊以及表頭。因爲這些封包是在區域内傳 送並沒有上到網路,所以任飪合適的封包格式都是可以使 用的,這包含了標準的與非標準的或是專有的格式。在本 較佳具體實施例中,使用的是專有的語音封包格式,此格 式與封包長度均針對該線路碼與線路技術,做出最佳化的 設計。該語音封包格式是由掛/離鉤,振鈴與切線信號 字,以及128位元之PCM»此語音封包更進一步地被封 裝成線路封包’其内部有又包含了位址、控制以及錯 誤更正位元》封包的總長度是1 4 〇位元。 現在,數據裝置112正透過乙太網路介面連結n3,將 乙太網路數據封包傳送至乙太網路控制器6 〇 5。曝後這些 -27- _本紙張尺度適用中國國家標準(CNS)A4規格(21G χ 297公爱) --- ---------------裝--- (請先閱讀背面之注意事項再填寫本頁) -線- 經濟部智慧財產局員工消費合、作社印製 4 6 6 86 4 A7 B7Λ7 4 6 β 8 6 4 ____ Η7__ 5. Description of the invention (23) The lower frequency of Hertz is dedicated to the use of POTTS. Three Plays generates a timing reference signal and passes it through the user loop 102 to the signal processor. The phase of the three-play timing reference signal is locked on the transmission link 203 and transmitted to the digital signal processor. The phase of the three-play timing reference signal is locked at the frequency of the transmission through the connection 203, and the digital signal is transmitted to the acquisition voice platform 10, the digital signal processor 501 and the acquisition voice data machine 1 frequency. It is specifically given to POTS. Transmission use. Sanwan generates a certain time reference signal and passes through the user circuit 102 to the relevant digital signal processor in 108. The phase of the three-play timing reference signal is locked on the network timing reference signal 8K_NTR transmitted to the digital signal processor 501 through the connection 203. In the illustration of the present invention, there is no difference between the three-play timing reference signal and the network timing reference signal 8K_NTR, and the network timing reference signal SK_NTR shown in the figure is from the voice acquisition platform 101 to the voice acquisition The modems 108 are non-interrupted. In the preferred embodiment, the Ethernet transceiver 5 007 is a 'Motorola MC68160 enhanced Ethernet serial transceiver; the processing tooth 50 2 is a Motorola model 1 ^ > 0860 £ 1 ^ power QUICC microprocessor; each digital signal processor 501 households use 'Texas Instruments model TMS320VC549 digital signal processor' to join Paradeen The software provided by the company, this digital signal processor can support the coding rules of the far-distance playing line code; each AFE 5 0 0 is adopted, the model of Abraham is AFE1137, fine-tuned to support the three-playing line code An analog front end for coding rules. -26-This paper size applies to Chinese National Standard (CNS) A4 (210 x 297 mm) (Please read the legal notices on the back and fill in this page first)--------- Order ---- ----- line. Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs ^ 6 6 864 A7 --------- B7___ V. Description of the Invention (24) Figure 6 shows the voice data acquisition device 1 of the present invention. 〇8 square. Block diagram. The telephone equipment 1 1 0 is connected to the subscriber line interface circuits (SLICs) 600 through a two-line connection u. The subscriber line interface circuit 600 receives the sharp ring analog signal from the telephone 11 (), samples the analog signal, and then uses the PC M technology to digitally encode the signal, and finally connects the 602 through the link 602. Wait for the speech samples to be transmitted to the processor 60. The sampling frequency is based on the 8-sh network timing reference signal 8K-NTR. The network timing reference signal 8K_NTR is transmitted by the digital signal processor 6 〇 4 to the subscriber line interface circuit 6 〇 3 through the link 6 〇 3. The sampling information is in the form of a 64-bit serial bit stream (including 8 samples with 8 bits per sample) per second 'from the subscriber line interface circuit 600 and transmitted to the processing through the link 602 The full-duplex serial communication channel interface (; § (: (; ^)) on the processor 601. The processor 60 1 decapsulates these PCM voice samples into voice packets, which contain signal information and headers. Because these Packets are transmitted in the area and not connected to the Internet, so any suitable packet format can be used. This includes standard, non-standard or proprietary formats. In this preferred embodiment, , Using a proprietary voice packet format, this format and packet length are optimized for the line code and line technology. The voice packet format is composed of hanging / off hooking, ringing and tangent signal words, And 128-bit PCM »This voice packet is further encapsulated into a line packet 'which contains address, control and error correction bits inside. The total length of the packet is 14.0 bits. Now, data devices 112 is going through B The network interface is connected to n3, and the Ethernet data packet is transmitted to the Ethernet controller 6 0. After exposure, these -27- _ this paper standard applies to China National Standard (CNS) A4 specification (21G χ 297 public love ) --- --------------- Equipment --- (Please read the notes on the back before filling this page) Printed 4 6 6 86 4 A7 B7
經濟部智慧財產局員工消費合作社印M 五、發明說明(25) 數據封包就會透過連結606 ’傳送至處理器601。若有必 要’處理器601會將該+乙太網路封.包分段成與該等語音 封包之大小相若。 然後’该等語音及乙太網路數據封包就會被緩衝在處理 器601中,透過匯流排607傳送至數位信號處理器6〇4中 之主處理器介面。場可程魂閘陣列6 〇 8在處迤器6 〇 r與數 位k號處理器6 0 4之間提供出膠邏輯。視所選元件不 同’可以不需要場可程式閘陣列6 〇 8。接.著,數位信號處 理器6 0 4會對該等語音及乙太網路數據封包執行以下的動 作:將其格式弄成串列的位元流型式,執行線路的編碼與 調變,以及透過連結6 1 0將該位元串流傳送至類比前端 609。類比前端609再*接著使用數位調變線碼,透過;結 109將該等位元驅動至用户迴路1〇2上!) ,爲確保此语音與數據封包混合傳送的方法可具有高的語 έι 質’處理器601賦予語音封包的傳送優先順序.,可保 註其上行及下行速率維持在每秒6 4仟位元β將非常小及 可比較的語音與數據封包大小組合,以及賦予語音封包高 於數據封包之傳送優先順序,就可以達到上$的要求。 下行的操作基本上是上述上行操作滅序的反推。語音封 包優先順序高於數據封包的執行是發生在局端侧,以使下 行封包可按其被接收的順序來接受處理。 在本較佳具體實施中,獲取語音數據機1〇8採用的是 通信系統公司所生產的型號爲Tg^100890_2C的超級線 路積體接取裝置。超級線路是該產品的註册商標。該超級 (請先閲讀背面之注意事項再填寫本頁) 裝Printed by the Intellectual Property Bureau's Consumer Cooperatives of the Ministry of Economic Affairs. V. Invention Description (25) The data packet will be transmitted to the processor 601 via the link 606 '. If necessary, the processor 601 will segment the + Ethernet packet. The packet is similar to the size of these voice packets. Then these voice and Ethernet data packets are buffered in the processor 601 and transmitted through the bus 607 to the main processor interface in the digital signal processor 604. The field programmable gate array 608 provides glue logic between the processor 600 and the digital k processor 604. Depending on the selected components, a field programmable gate array 608 may not be required. Next, the digital signal processor 604 will perform the following actions on these voice and Ethernet data packets: format its format into a serial bit stream pattern, perform line encoding and modulation, and The bit stream is transmitted to the analog front end 609 via link 6 1 0. The analog front end 609 * then uses a digital modulation line code and transmits through; the end 109 drives these bits to the user loop 102! ), In order to ensure that this method of mixed transmission of voice and data packets can have a high voice quality, the processor 601 gives priority to the transmission of voice packets. It can ensure that its upstream and downstream rates are maintained at 64 bits per second. Beta combines very small and comparable voice and data packet sizes, and gives voice packets a higher priority than data packets for transmission. The downlink operation is basically the reverse deduction of the above-mentioned uplink operation. Voice packets have a higher priority than data packets. The execution takes place at the office side so that downstream packets can be processed in the order in which they were received. In the preferred embodiment, the voice data acquisition device 108 uses a super-line integrated access device of the type Tg ^ 100890_2C produced by the communication system company. Super Line is a registered trademark of this product. The super (Please read the notes on the back before filling this page)
If ----訂---------線' J- 本紙張尺巾_家鮮(CNS)A域格(21G X 297公楚"j 4 6 6 86 4 Λ7 B7 五、發明說明(26) 線路轉接器是根據帕拉代恩公司的三玩技術所設計出來 的。 從圖6中尚可看到連接至用户迴路1〇2之p〇TS電話 1 1 4。標準POTS服務乃與本發明之獲取垮音線路共享用 户迴路102。本較佳具體實施之低通濾波器116其功用在 於,將POTS電話1〗4與該較高頻的獲語音線路信號相隔 開。. 其他的具體實施例 雖然本發明之較佳具體實施例已説明完畢,但在以下專 ,利範儔内之其他方式也是可以實施的。欲以下列之專利及 他們的等同物來界定的發明範圍。 任何牽涉到將本發明予以修改的實施方式,以符合物理 上的、技術上的或經濟上的限制或是客户的需求,只要不 脱離本發明之範圍與精神,都是可允許執行的。譬如,電 話設備的數量以及獲取語音數據機108可支援的數據裝置 數,均受限於連結109與用户迴路102可提供出之總傳輸 頻寬,每一個現用狀態的電話設備1 1 0所獨占去的頻寬, 以及傳輸數據時可接障的最小頻寬的限制。再者,這些頻 寬參數也與所選傳輸技術與相關協定,以及用户迴路狀況 有關β —般而言,獲取語音電話設備110的數量與獲取語 音數據機108可支援的電腦裝置數,獲取語音平台1〇1可 支援的獲取語音數據機108的數量,每個獲取語音平台 101可接的線路卡200的數量,以及每個線路卡200可接 -29- 本纸張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐) C請先閱讀背面之注意事項再填寫本頁) € 裝--------訂·! 線. 經濟部智慧財產局員工消費合作杜印製 4 6 6 86 4 A7 ___]£ 五、發明說明(27 ) 的川户竹路丨〇 3的數量,在設計時均受到上述因素的限 制,而依本發明苑丨,¾可涵蓋多枝的數·g: » 阑7與8分別地领示如何於單一用户迴路上,規劃支援 單一電話設備與數據裝冱之語音卡及數據卡。圖7與8之 元件所執行的基本功能,與上述一般化之多數據裝置與電 話設備中之元件所執行的功能相同。若僅有一個電話設備 且沒有數據裝置,就不需要數據處理器卡2(Π,乙太網路 控制器3 0 0就可直接地連接至乙太網]路收發器5 〇 7。另 外’透過八條線來連接用户迴路〗〇 3之類比前端5 〇 〇與數 位信號處理器501也不需要了 β 二當有需要多接用户迴路! 〇3時,才會需要相對應的類比 七祝500科數位旮號處理器5〇1。當有額外的線路卡出現 時’才有可能需要額外的乙太網路交換器4GG來作爲 的命令器。 同樣地’牵涉到特殊元件選擇之具體實施方式,以符合 物理上的、技術上的或經濟上的限制,只要不脱離本發明 (範圍與精神’都是可允許執行的。譬如,從一個工廠中 所生產出來之元杜 社古t 經 濟 部 智 慧 財 產 Μι 員 工 消 費 合 作 製 的膠邏輯。心所要本具體實施例所指出 具(或更不具)集成性凡件有可能較其他的選〜更 擇,在”昤、/ (或更不具)能力。元件的選 蓋許多的到上述因素的限制,而本發明範圍可涵 之上,傳送該語音及數據封包 馬,乃本發明所附帶出來的。具下列 I_______ w ου - 4 6 6 Λ7 Ιί7 五、發明說明(28) 能力的數位調變線路碼都是可以使用的:在同一個實體用 户迴路上’其載波頻帶可與其他服務(像是,p〇TS )的頻 帶相隔離;衧足夠的频寬支援所希望擁有的電話設備與數 據裝冱數;功率頻譜密度在聯邦通訊委員會的规範與調節 範園之内;以及有能力傳輸網路定時參考信號。譬如,國 際電信聯盟所製定的標準:G.lite就是另一種可接受的數 位調變線路碼。 相類似的,進入該數據網路用之傳輸協定與連結i 〇 6, 本具體實施例所示之爲乙太網路介面連結,但其實可以是 任何合適的協定以及可滿足應用特別需要之實體連結。 在本較佳具體實施例中,後平面202乃是使用乙太網路 協定,來作線路卡200 '數據處理器卡2〇1以及語音處理 處理器卡204間,語音及數據封包的傳送。不過,其他一 些以協定爲基礎之細胞和架構也是可以使用。譬如,也可 以使用具A T Μ交換功能之ATM25。 本較佳具體實施例所描述之系統,其中獲取語音平台 乃疋過中繼器105 ’直接地與局端交換機1〇4連 接,另外,也透過連結106,直接地與數據網路連接。習 於此藝人士需知道電信網路可能牵涉到許多的網路元件, 只是要在本發明之範圍内將介面相關的事物作改變,就可 以將獲取語音平台1 〇 1做進網路中任何的分歧點内。 語音及數據封包的大小,乃取決於設計時所選的基本協 定與傳送技術,且必須與傳送語音的最小所需時間有所平 (請先閱讀背面夂注意事項再填寫本頁) ί I ί — I I 訂--! I ] I [ [ I _ 經濟部智慧財產局員工消費合作社印製 -31 -If ---- Order --------- Line 'J- This paper ruler_ 家 鲜 (CNS) A domain grid (21G X 297 Gongchu " j 4 6 6 86 4 Λ7 B7 V. Description of the Invention (26) The line adapter is designed based on Paradion's Sanwan technology. From Fig. 6, it can be seen that the POTS telephone 1 1 4 is connected to the subscriber circuit 102. Standard The POTS service shares the user circuit 102 with the acquired collapsed line of the present invention. The function of the low-pass filter 116 of this preferred embodiment is to separate the POTS phone 1 from the higher frequency received voice line signal. Other Specific Embodiments Although the preferred embodiments of the present invention have been described, other methods in the following patents and methods can also be implemented. The scope of the invention is to be defined by the following patents and their equivalents. Any implementation involving modification of the present invention to meet physical, technical, or economic constraints or customer needs is permissible as long as it does not depart from the scope and spirit of the present invention. . For example, the number of telephone equipment and access to voice modem 108 can support The number of data devices is limited by the total transmission bandwidth provided by the connection 109 and the user circuit 102, the bandwidth exclusively occupied by each active telephone device 110, and the minimum frequency that can be blocked when transmitting data. Broadband restrictions. In addition, these bandwidth parameters are also related to the selected transmission technology and related agreements, as well as the condition of the user's circuit. Generally speaking, the number of voice telephone devices 110 and the computer devices supported by the voice modem 108 can be obtained. Number, the number of voice modems 108 supported by the voice platform 101, the number of line cards 200 that each voice platform 101 can access, and the number of line cards 200 that can be accessed by each line card -29 China National Standard (CNS) A4 specification (210 X 297 mm) C Please read the precautions on the back before filling in this page) € Install -------- Order ·! The consumer cooperation of the Intellectual Property Bureau of the Ministry of Economic Affairs, Du printed 4 6 6 86 4 A7 ___] £ V. The number of Kawado Bamboo Road 丨 〇3 of the invention description (27) is limited by the above factors when designing, According to the present invention, the number of branches can be covered by g: »The laps 7 and 8 respectively show how to plan a voice card and a data card that support a single telephone device and data device on a single user circuit. The basic functions performed by the components of Figs. 7 and 8 are the same as the functions performed by the components of the generalized multi-data device and telephone equipment described above. If there is only one telephone device and no data device, there is no need for a data processor card 2 (Π, Ethernet controller 3 0 0 can be directly connected to the Ethernet] road transceiver 5 07. In addition ' The user circuit is connected through eight lines. 〇3 analog front end 500 and digital signal processor 501 do not need β. Second, when there is a need to connect more user circuits! 〇3, the corresponding analog 7 will be needed. 500 digital digit processor 501. When there is an additional line card, it is' possible to require an additional Ethernet switch 4GG as a commander. Similarly, it involves the implementation of special component selection. In order to comply with physical, technical, or economic constraints, as long as it does not depart from the present invention (the scope and spirit 'are permissible. For example, Yuan Du She Gu produced from a factory Intellectual property of the Ministry of Economic Affairs, the consumption logic of the cooperative system of employees. What I want This specific embodiment indicates that the integrated (or not) integrated components may be better than other options. )can The selection of components covers many of the above factors, and the scope of the present invention can be covered. The transmission of the voice and data packets is attached to the present invention. The following I_______ w ου-4 6 6 Λ7 Ιί7 5 2. Description of the invention (28) Digitally modulated line codes with capabilities are available: on the same physical user circuit, its carrier frequency band can be isolated from the frequency bands of other services (such as p0TS); Bandwidth supports the desired number of telephone equipment and data devices; power spectral density is within the Federal Communications Commission's specifications and regulation paradigm; and the ability to transmit network timing reference signals. For example, standards developed by the International Telecommunication Union : G.lite is another acceptable digital modulation line code. Similarly, the transmission protocol and connection i 〇6 used to enter the data network are shown in this specific embodiment as an Ethernet interface connection. But in fact, it can be any suitable protocol and physical connection that can meet the special needs of the application. In this preferred embodiment, the rear plane 202 is an Ethernet protocol. It is determined that voice and data packets are transmitted between the line card 200 'data processor card 201 and the voice processing processor card 204. However, other protocol-based cells and architectures can also be used. For example, also ATM25 with AT MM switching function can be used. In the system described in this preferred embodiment, the voice platform is directly connected to the central office switch 104 via the repeater 105 ', and also through the link 106 Connect directly to the data network. Those skilled in the art need to know that the telecommunications network may involve many network components, but to change the interface-related things within the scope of the present invention, the voice platform can be obtained. 1 〇1 into any points of divergence in the network. The size of the voice and data packets depends on the basic protocol and transmission technology selected at the time of design, and must be equal to the minimum time required to transmit the voice (please read the precautions on the back before filling this page) ί I ί — II Order-! I] I [[I _ Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs -31-
6 6 8 6 4 λ7 ____w___ 五、發明說明(29) 衡。不同的技術會產生不同的絕對及相對的語音及數據封 包尺寸。 , 本應用也可以相位法來實施i此法一開始就提供有額外 的語音線路服務,而只在中央機房的數據傳輸底層建立好 之後,才提供有數據服務。 (請先閱讀背面之注意事項再 .袭--- 承寫本頁) Ή. 線. 經濟部智慧財產局員工消費合作社印製 -32- 本紙張尺度適用中國國家標準(CNS)A4規格(210 X 297公釐)6 6 8 6 4 λ7 ____w___ V. Description of the invention (29) Balance. Different technologies produce different absolute and relative voice and data packet sizes. This application can also be implemented by the phase method. This method provides additional voice line services from the beginning, and provides data services only after the bottom layer of the data transmission in the central computer room is established. (Please read the precautions on the reverse side first. --- Write this page) Ή. Thread. Printed by the Consumer Cooperatives of the Intellectual Property Bureau of the Ministry of Economic Affairs-32- This paper size applies to China National Standard (CNS) A4 specifications (210 X 297 mm)
Claims (1)
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
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US09/398,697 US6747995B1 (en) | 1998-09-21 | 1999-09-20 | System for multiple voice lines with data over a single subscriber loop |
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TW466864B true TW466864B (en) | 2001-12-01 |
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ID=23576431
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TW89106612A TW466864B (en) | 1999-09-20 | 2000-04-10 | System for multiple voice lines with data over a single subscriber loop |
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Cited By (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
TWI427977B (en) * | 2009-04-17 | 2014-02-21 | Hon Hai Prec Ind Co Ltd | Voip device and method of reducing noise thereof |
DE202015102010U1 (en) | 2014-09-19 | 2015-06-01 | Yuan-Hung WEN | Quick release connector |
-
2000
- 2000-03-27 CA CA002302127A patent/CA2302127C/en not_active Expired - Fee Related
- 2000-04-10 TW TW89106612A patent/TW466864B/en not_active IP Right Cessation
Cited By (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
TWI427977B (en) * | 2009-04-17 | 2014-02-21 | Hon Hai Prec Ind Co Ltd | Voip device and method of reducing noise thereof |
DE202015102010U1 (en) | 2014-09-19 | 2015-06-01 | Yuan-Hung WEN | Quick release connector |
Also Published As
Publication number | Publication date |
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CA2302127A1 (en) | 2001-03-20 |
CA2302127C (en) | 2007-06-26 |
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