Dec 10, 2007

CNNIC:IPv4可能提前耗盡 漲價趨勢日益明朗

發佈時間:2007.12.06 17:04 來源:DoNews 作者:DoNews

近日,中國互聯網路資訊中心(CNNIC)發佈了IP地址最新動態:根據IPv4地址消耗速度,IPv4地址將在2011年左右耗盡的預測可能會提前到來。同時,因為IPv4嚴重緊缺,相關國際組織擬採用澳元形式收費,由於匯率的原因,IPv4地址價格上揚已成定局。

IPv4枯竭可能提前到來

“IPv4將於2011年左右耗盡的結論並非聳人聽聞,而是使用國際通行的IPv4地址消耗預測法得出的結論,這一預測法根據全球IPv4地址的消耗速度進行函數預測。CNNIC從1998年開始做這樣的預測以積極應對IPv4的‘入不敷出’。” CNNIC IP地址專家李祥建告訴記者。

比較我國IPv4預測消耗與實際消耗,我們明顯看出:‘入不敷出’的現象已經頻頻發生——2006年我國IPv4地址預測數量約為9500萬,實際消耗量為9800萬,超出預測近300萬;2007年,IPv4預測量為1.05億,截至10月27日,我國IPv4實際數量已經達到1.31億,偏差超過2600萬。對此,李祥建指出:“IPv4將於2011年左右耗儘是根據幾年前的消耗速度做出的預測,按目前的情況來看,耗盡很有可能提速,不用等到2011年IPv4就會枯竭。”

CNNIC的調查數據再次印證了我國IPv4的激增:2005年,我國IPv4數量為7400萬個,2006年底,我國IPv4數量為9800萬個,年漲幅為32%;截至10月27日,我國IPv4數量已達13189萬個,排名世界第三,年漲幅飛升至77.2%。

IPv4地址嚴重緊缺, IPv4地址漲價趨勢在所難免,相關國際組織已經開始積極調整IP地址價格和收費方法以適應IP地址分配的發展形勢。據悉,2008年1月開始,亞太地區IP地址申請將採取澳元收費,因為澳元利率近年保持持續高漲趨勢,這一收費方式意味著我國IPv4地址的實際申請價格上也將隨之上揚。

“兩手抓”應對IPv4 地址荒

IPv4將於2011年耗盡,地址漲價已成定局,成為擺在我國廣大ISP面前的一個“燙手山芋”。李祥建告訴記者:“IPv4 緊缺導致的漲價將直接造成兩個惡性迴圈:一個是‘囤積居奇’的錯誤申請思路,對IPv4進行了大規模非預期申請,缺乏宏觀調控 ,從而加速IPv4的耗盡;第二個是“節衣縮食”的錯誤使用方法,出於對IPv4耗盡的恐慌緊縮IPv4的使用,建立在此基礎上的網路應用隨之受阻,抑制中國互聯網整體發展。

業界專家普遍認為: IPv6是“根治”這兩個惡性迴圈的良藥,但目前這劑良藥只有藥方並未製成,亦無法大面積解決“IPv4耗盡之痛”,IP地址耗盡的2011年前我們應該怎麼辦?

不久前,CNNIC發佈應對IP地址枯竭的聲明,為ISP們提供了“暫緩病痛”的方法:依託其在國際、國內互聯網社區的地位與相關組織的積極協調,在亞太地區尤其是中國建立起應對IPv4地址資源耗盡問題的共識,保證在過渡時期IPv4地址分配和管理一致。

據李祥建解釋:“這一‘共識’便是雙管齊下的申請策略,一方面積極申請足量的IPv4地址,以應對IPv6‘遠水不解近渴’的窘局;另一方面努力備戰未來的IPv6地址建設,以求徹底‘根治’IPv4 地址荒。”據悉,目前全球剩下10億個左右IPv4地址可用,而且因為歷史原因分配給美國而閒置的14.2億個IPv4地址,在國際互聯網組織的努力下正被逐漸釋放出來以備利用。CNNIC的這一聲明對於廣大ISP無疑是一顆“定心丸”——IPv4地址仍有較大迴旋空間,不必因為IP地址的青黃不接而自亂方寸。

另一顆“定心丸”則是:作為中國的IP地址分配管理機構, CNNIC 的IP地址分配窗口已達到4B,成為世界最大的分配窗口,能一次性自主分配26萬多個IP地址,這意味著CNNIC的IP地址分配能力已居世界首位。目前CNNIC已累計為國內廣大ISP分配地址3672萬個。


(責任編輯:燕山)

Dec 6, 2007

To Be or Not to Be a Instructor ?

要成為一個instructor需要的不只是經驗及技術,還要有一股熱忱,因為當你面對的不再是冷冰冰的設備時,你必須要面對的是各式各樣的學生,如果你無法從教學的過程中得到自我滿足及成就感,相信我,這份工作比當工程師辛苦好幾倍以上(這僅是以個人在ISP網路工程師的經驗而言,如果是工程師的話,白天上班時間只要把自己該作的事情作好,剩下的就是你自己可以利用的時間;但是如果是當講師的話,從早上講到吃晚飯,辛苦一點的話,有可能還有晚上班從吃完晚飯之後開始講到你準備上床睡覺…如果再苦命一點,假日班也要上課的話,幾乎沒有所謂的上班偷閒的時間,甚至也沒有了所謂的假日)。所以如果你沒有體力之外可以支撐你的精神力量,可能無法長久持之以恆下去。

再來談到另一個問題,那就是課程的準備及設備的架設,不暪您說,以小弟目前的工作而言,通常每個星期都必須要花費至少半天或是一整天的時間準備下一個星期的設備架設;若是要教導一課新課程,至少利用每天下班時間研讀兩個星期以上,先準備完筆試之後,再準備嘗試自己架設Lab的環境。

說到這邊,可能就會有人問為什麼這麼辛苦的工作,為什麼還是會有人要作? 一個原因是相對於一般工程師來說,講師的時薪或許是比較高的;另一個原因就是你真的可以在教學的過程進而將你所教授的課程真正地融會貫通,而不單單只是了解大概內容;最後一個原因應該會是絕大部份資深講師的共同特點,他們都樂於與人分享,為人解惑,在課堂中教學相長的感覺是很難用言語形容的,這個原因對部份的人來說可能比高薪更令他們嚮往。

最後還有一個當初小弟也一直很擔心的問題,那就是成為教育事業的一份子之後,是否會與實際的網路工作脫節,是否會無法追上現在各種新技術的腳步。這個問題,TP的前輩給了我一個很好的答案:"也對,也不對!" 當你離開網路公司/系統整合公司之後,你真的很難再有機會隨時將你所學在實際的網路中去進行測試及整合,但是,不代表你會無法追上新技術潮流…因為你必須不斷地學習,無論是上班時的教學相長或是下班之後的自修,你將會從學生那邊的互相討論中領會到現今技術的應用及客戶的技術需求及問題,也會為了準備未來的新課程或是舊課程改版,你勢必要比其他學生更早接受新教材或是自己要先想辦法了解新技術的來龍去脈,所以簡單一句話,當你成為講師時,你將無時無刻不在進修,跟其他一般工程師忙於routine工作或是為了應付客戶而虛耗時間比較起來的話,進步的速度將無法比擬(僅個人經驗,過去一年我所學得的東西超過我當工程師三年內所學的總和…不過真的都只有紙上談兵及Lab環境的測試,無法再跟現實production網路經驗相提並論)

總而言之,要當講師必須要有一個心理準備,那就是你必須真心問自己你是否真的好學? 你真的喜歡每天把唸書當工作? 是否對所有新事物都有一股好奇心?(我現在的生活很像準備大學聯考的高中生,工作的內容就是講課/聽課(這是當講師最大的好處)/看書/查資料(教材的內容永遠不夠)/考試(當講師的另一個好處,不用再自掏腰包去考試了,但是相反地現在會有人來逼你…不得不去考試))。你必須要自己調整學習步伐配合公司的目標及每個月的開課內容規劃,在學習技術的同時還要學習如何表達自己的學習祕訣/讀書心得/個人想法/課程重點/工作經驗,努力地嘗試去符合不同學生的需求與期待,利用學生可以接受的各種方式來授課。

這些問題也是過去小弟轉換跑道時很想了解的內容,趁此機會把個人經歷在此分享,也是小弟過去這段日子以來的工作感想,僅供給各位參考!

Dec 1, 2007

【好片推薦】不能說的祕密

昨天終於把CVOICE全面看完,該準備的補充資料也備齊了,該考的筆試也過了,所以終於可以喘口氣,原本準備在睡前把Cisco - Voice over IP Fundamentals看完之後就上床,不過因為看完之後才PM 11:00所以想說把之前堆積一堆電影看一下給自己放鬆一下。

挑來挑去就選了這一片好評不斷的電影 - 不能說的祕密。網路有一個好處,那就是可以根據大多數的人意見整合出一個很客觀的結果(雖然這個結果是來自於大多數人的主觀),所以我通常是根據這個指標來決定我下一部想看的影片。



"不能說的祕密" - 從這部影片的名稱看來其實可以猜到一二,一定跟愛情有關,因此原本有抱持著準備看連續劇的想法來看這部影片,不過片中的男女主角的感情發展說實在腳步進行的很快(不過這搭配上劇情來說也是合理,如果你到了另一個世界,只有一個異性能跟你溝通,只有這個人能夠知道你的存在,大概也會是這樣的情況)。

整部影片拍攝的感覺很好,漸漸脫離了港式江湖味,也少了台灣傳統國片的沈悶,但多了一點韓片的精緻及劇情細膩安排(也有人批評有抄襲之嫌,不過我想如果單以這部電影的劇情及結局安排來說,比韓片的悲情安排或刻意的喜劇收場來得特別)雖然角色不多,但是每個人的情緒及表現都溶入劇情中,雖然有些新人演技有點生澀,但無礙於整個劇情流暢的演進,更有趣地是常見的一點一滴漸漸揭開觀眾心中的問題之倒述法,令不少人在觀看電影的過程中猜測這是一部靈異片(就好似最近的一部西片-隱形人),基本上"不能說的祕密"應該要算是少見的科幻浪漫愛情國片。

看完這部片我個人最好奇的事情跟大部份人一樣那就是"結局"的合理化。由於只用了一張畢業照來帶過,因此給人很大的想像空間,這一點是我個人最喜愛的部份。在網路上討論最多的莫過於大家發表各自對於這樣結局的安排的劇情猜測,這個地方尤其讓人玩味的就是周杰倫出現在桂綸鎂的旁邊,但是許多人完全沒有注意到他的存在,因為…還蠻不像的,少了那個酷酷的感覺,變成一個靦腆的男學生。我看過周杰倫的頭文字D那部影片,說真的演技蠻爛的…但是從這部片中我感覺到他的改變,開始有電影男主角的感覺了,…至少話變多了雖然還是維持他的特色,每一段話都不長,但是多了許多的幽默的對白,這要感謝編劇的努力,不然這部片會少了一些笑聲。

其中的女主角小雨(桂綸鎂)及女配角(晴依)曾愷玹我覺得很棒,雖然對於曾愷玹來說戲份不多,但是曾愷玹的個人特質不錯,清新而亮麗,長相也很討喜,我想未來一定會漸漸發光。桂綸鎂就不用談了,很多人因為這部片喜歡上了桂綸鎂,包括她的下一部片-最遙遠的距離,已經有很多影迷拭目以待。我個人覺得桂綸鎂的演技一流,完全將女主角的特別表露無遺再加上個人的詮釋給予這個角色一些不同於劇本的特殊感覺,我想未來的星途不可限量,只要有好的劇本及導演的賞識,應該會有更棒的表現。

現在來談談這部片的結局,很多人覺得不合理的是為什麼周杰倫回到了過去,但是卻能出現在畢業照中,也就表示了在過去的時光中,周杰倫是可以被所有的人看得到的。個人的想法是劇情的安排,如果在琴房中正常的演奏琴譜,將可以直接跨越時空來到20年後的今天,但是將只會被20年後第一眼看到人的發現自己的存在;如果在琴房中以快速的節奏來彈琴,那麼就會直接回到20年前的今天(不過這邊留下了一個小小伏筆,似乎並不一定是同一天的同一個時間點,有可能比較早或是比較晚,這樣才有辦法讓周杰倫回到過去將桂綸鎂從無解的愛情中解救出來)。

「我閉上眼睛,是為了看清楚你。」這句話寫得很深清,更將影片中的重點深藏其中,唯有看過電影的觀眾才會心有戚戚焉,當電影散場之後各位再重新思考電影海報上這一句話,你會有一種大夢初醒的感覺。我的想法是如果從過去來到未來的規則中,是只能被第一眼看到的人所發現的,而其他人則完全視若無睹;但是如果是從未來回到過去的話,那麼就是一切正常,也就是將可以讓所有人的看見自己的存在。這樣的推論就可以讓周杰倫為何可以跟桂綸鎂一同出現在大家都可見的畢業照中了。只是比較沒有直接的證據可以佐證的是周杰倫回到的過去時間點到底是在桂綸鎂發現不可思議的琴譜之前或之後?:

  • 有人說是在桂綸鎂死前的那一天,看起來桂綸鎂並不認識周杰倫,事實上是因為桂綸鎂的調皮刻意假裝的…。不過這一點比較難以說明為何桂綸鎂會正常在學校教室中,因為桂綸鎂在氣喘發作當天之前五個月都沒有回到學校中(也沒有再去琴房跑到未來去找周杰倫,更不可能會乖乖地待在教室中寫東西。
  • 也有人說是在桂綸鎂發現琴譜之前,因此真的完全不認得周杰倫了,只是因為周杰倫的樣子讓她不自主地的發笑了。(這樣的推論是比較合理的。但是時間點的問題真的無法推論,因為如果可以透過彈琴的節奏來控制回到過去的時間點,可能會發生同一個時間點同時出現兩個一樣的人,所以…請自己合理化吧!)
  • 也有人說事實上周杰倫並來不及按下最後一琴鍵就已經死了,後面的劇情只是周杰倫死前最後的想像情節。(這也許是影迷最不希望的結局)
  • 個人比較喜歡的結局是周杰倫回到了桂綸鎂發現琴譜之後,已經認得周杰倫了,但是是在桂綸鎂撞見女配角跟周杰倫在琴房親嘴那一幕之前,所以桂綸鎂還是一貫地調皮刻意假裝不認得來捉弄周杰倫。(不過有點小小不合理啦,因為若真是如此,按理來說桂綸鎂的表現應該是驚喜不已才是,沒有想到周杰倫真的發現了這個祕密並且回到過去來找她,桂綸鎂要能控制住自己情緒還來捉弄周杰倫實屬不易…)



不論如何,這是一部成功的國片,尤其是片中的動畫場景真的已有國際性的水準,像是當桂綸鎂第一次演奏那份琴譜時,週遭事物的變化跟洋片的一些時間穿梭片段影像處理水準不相上下。雖然本片沒有像色戒中大卡司及名導演的加持,不過個人還是給予這部片極高的評價,只希望新生代的演員可以突破舊有國片惡劣的環境,重新讓國片受到國人及國際間的注視!

Nov 29, 2007

PCM vs ADPCM vs CS-ACELP vs LD-CELP

一旦我們的類比波形已經被數位化之後,我們可能想要透過編碼這些數位化波形來加以壓縮好節省廣域網路的頻寬。編碼和解碼這些波形的過程是由編解碼器(coder decoders),也被稱之為編解碼器(code decoder,CODEC)。讓我們來看看各種的編解碼器(CODECs)所使用的波形壓縮的一些形式︰

脈衝編碼調變(Pulse Code Modulation, PCM)
是將類比訊號轉換為數位訊號的一種技術,它並非實際地壓縮類比波形。相反的,脈衝編碼調變(PCM)取樣以及數位量化的動作並未進行任何的壓縮。G.711就是使用脈衝編碼調變(PCM)的編解碼器(CODEC)。

可適性差分脈衝碼調變(Adaptive Differentiated PCM,ADPCM)
使用一個差異化訊號(difference signal)。不將整個樣本加以編碼,可適性差分脈衝碼調變(ADPCM)可以將目前樣本與前一個樣本比較出來的差異傳送出去。
G.726就是一個可適性差分脈衝碼調變編解碼器(ADPCM CODEC)的例子。

代數碼激勵線形預測(Conjugate Structure Algebraic Code Excited Linear Predication,CS-ACELP)
動態建造基於語音模式的編碼登錄(codebook)。它會使用一個前看緩衝區(look ahead buffer)來查看是否下一個樣本與已存在於編碼登錄(codebook)中的圖案相配。如果它這樣做,那麼編碼登錄(codebook)位置就可以被傳送出去來取代實際的樣本。這個好處就是不需要傳送實際的聲音,我只要傳送給您那個聲音在您編碼登錄(codebook)中的位置,相較於傳送實際的聲音使用了相當少的頻寬。
G.729就是一個代數碼激勵線形預測編解碼器(CS-ACELP CODEC)的例子。

低時延碼激勵線性預測(Low-Delay Conjugate Excited Linear Predication,LD-CELP)
非常類似於代數碼激勵線形預測(CS-ACELP)。不過,低時延碼激勵線性預測(LD-CELP)使用了更小的編碼登錄(codebook),導致較少的延遲,但是它需要更多的頻寬。
G.728編解碼器(CODEC)是一個低時延碼激勵線性預測編解碼器(LD-CELP CODEC)的例子。

G.729所使用的前看緩衝區(look ahead buffer)的目的是為了在一個緩衝區中收集語音模式並且試圖將這些語音模式與已經存在於本地編碼登錄(codebook)中的模式來配對。事實上,在思科網路電話(VoIP環境)中,G.729是在廣域網路中最受歡迎傳送語音流量的編解碼器(CODEC),主要是因為它的高品質及低頻寬需求。為了傳送實際數位化的語音,G.729 只需要8 kbps,相較於G.711所需要的64 kbps頻寬。代數碼激勵線形預測(CS-ACELP)被設計用來把語音模式加以編碼。因此,其他音頻的來源(例如,來電等待音樂)與人類講話相比較,可能會經歷到更多的品質降低的情況。

您通常將會在本地區域網路環境中使用G.711(需64 kbps的頻寬來支付負荷語音流量)而在廣域網路環境中使用G.729(需8 kbps的頻寬來支付負荷語音流量)。所有種類的G.729需要8 kbps頻寬來傳送語音,以下是兩種G.729變型的差異之處:
  • G.729a使用一種較不複雜的演算法,保留處理器的資源伴隨著非常輕微的品質降低(quality degradation)。
  • G.729b支援語音活動偵測(voice activity detection,VAD)。

Note: 語音活動偵測(VAD)可以偵測談話什麼時候停止。在思科路由器上預設情況下,在250毫秒(ms)的寂靜(silence)之後(意即,4分之1秒),路由器停止傳送寂靜(silence),因此釋放出可用的頻寬。雖然語音活動偵測(VAD)所節省的頻寬數量基於語音模式變化而不同,但是通常可以節省百分之35的頻寬。

Understanding How Digital T1 CAS (Robbed Bit Signaling) Works in IOS Gateways

Channel Associated Signaling (CAS) is also referred to as Robbed Bit Signaling. In this type of signaling, the least significant bit of information in a T1 signal is "robbed" from the channels that carry voice and is used to transmit framing and clocking information. This is sometimes called "in-band" signaling. CAS is a method of signaling each traffic channel rather than having a dedicated signaling channel (like ISDN). In other words, the signaling for a particular traffic circuit is permanently associated with that circuit. The most common forms of CAS signaling are loopstart, groundstart, Equal Access North American (EANA), and E&M. In addition to receiving and placing calls, CAS signaling also processes the receipt of Dialed Number Identification Service (DNIS) and automatic number identification (ANI) information, which is used to support authentication and other functions.

Each T1 channel carries a sequence of frames. These frames consist of 192 bits and an additional bit designated as the framing bit, for a total of 193 bits per frame. Super Frame (SF) groups twelve of these 193 bit frames together and designates the framing bits of the even numbered frames as signaling bits. CAS looks specifically at every sixth frame for the timeslot's or channel's associated signaling information. These bits are commonly referred to as A- and B-bits. Extended super frame (ESF), due to grouping the frames in sets of twenty-four, has four signaling bits per channel or timeslot. These occur in frames 6, 12, 18, and 24 and are called the A-, B-, C-, and D-bits respectively.

The biggest disadvantage of CAS signaling is its use of user bandwidth in order to perform signaling functions.

Super Frame is an older framing standard for T1s. Also called D4 or D3/D4 framing. In the 1970s it replaced the original T1/D1 framing scheme of the 1960s in which the framing bit simply alternated.

In order to determine where each channel is located in the stream of data being received, each set of 24 channels is aligned in a frame. The frame is 192 bits long (8 * 24), and is terminated with a 193rd bit, the framing bit, which is used to find the end of the frame.

In order for the framing bit to be located by receiving equipment, a pattern is sent on this bit. Equipment will search for a bit which has the correct pattern, and will align its framing based on that bit. The pattern sent is 12 bits long, so every group of 12 frames is called a Super Frame. The pattern used in the 193rd bit is 1000 1101 1100.

Superframe remained in service in many places through the turn of the century, replaced by the improved Extended Super Frame of the 1980s in applications where its additional features were desired.

In telecommunication, an Extended Super Frame (ESF) is a T1 framing standard, sometimes called D5 framing, invented in the 1980s. It is preferred to its predecessor, Super Frame, because it includes a cyclic redundancy check (CRC) and bandwidth for a data link channel (used to pass out-of-band data between equipment.) It requires less frequent synchronization than the earlier superframe or D-4 format, and provides on-line, real-time testing of circuit capability and operating condition.

In ESF, a superframe is 24 frames long, and the 193rd bit of each frame is used in the following manner:

  • Frames 4, 8, 12, 16, 20, and 24 are used to send the framing pattern, 001011
  • Frames 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21 and 23 are used for the data link (totalling half of all framing bits, or 4000 bits per second)
  • Frames 2, 6, 10, 14, 18, and 22 are used to pass the CRC total for each super frame.

Note: Less-frequent synchronization frees overhead bits for use in testing and monitoring.

CAS Signaling Types

Loopstart Signaling
Loopstart signaling is one of the simplest forms of CAS signaling. When a handset is picked up (the telephone goes off-hook), this action closes the circuit that draws current from the telephone company CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the telephone to ring.

A disadvantage of loopstart signaling is the inability to be notified upon a far-end disconnect or answer. For instance, a call is placed from a Cisco router configured for Foreign Exchange Station (FXS)-loopstart. When the remote end answers the call, there is no supervisory information sent to the Cisco router to relay this information. This is also true when the remote end disconnects the call.

Note: It is possible for answer supervision to be provided with loopstart connections if the network equipment can handle line-side answer supervision. Also, loopstart provides no incoming call channel seizure. Therefore a condition known as glare can arise, where both parties (Foreign Exchange Office [FXO] and FXS ) try to simultaneously place calls. Glare can be avoided when you configure the T1-CAS gateway's port selection order in such a way that the inbound and outbound calls are in reverse order. For example, if the inbound calls are sent by the provider on the FXO ports in the order of port 1, port 2, port 3 and port 4, then configure the Cisco CallManager Route Group to route outbound calls on those same ports in the order port 4, port 3, port 2 and port 1.

With loopstart signaling, the FXS side only uses the A-bit and the FXO side only uses the B-bit to communicate call information. The AB-bits are bi-directional. This state table defines this signaling information from the CPE's perspective (FXS).

This is the FXS-loopstart timing diagram.


On an incoming call (network -> CPE) this happens:

  1. The network toggles the B-bit to indicate ringing. This is a standard ringing pattern. For instance, 2 seconds on, 4 seconds off.
  2. CPE detects the ringing and off-hook states. A-bit goes from 0 to 1.

In an outgoing call (CPE -> network) this happens:

  1. CPE goes off-hook and A-bit goes from 0 to 1.
  2. The network provides dial tone. There is no signaling change.
  3. CPE sends digits (dual tone multifrequency (DTMF) in Cisco's case).

During a disconnect from the network, this occurs:

  1. CPE detects in-band that the call has dropped (someone says good-bye or a modem drops the carrier).
  2. CPE goes on-hook and A-bit goes from 1 to 0.

During a disconnect from the CPE, only step 2 occurs.

The Answer Supervision and Disconnect Supervision States are only seen when provided by the network.

Groundstart Signaling
Groundstart signaling is very similar to loopstart signaling in many regards. It works by using ground and current detectors that allow the network to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for positive recognition of connects and disconnects. For this reason, ground start signaling is typically used on trunk lines between PBXs and in businesses where call volume on loop start lines can result in glare.

The advantage of groundstart signaling over loopstart signaling is that it provides far-end disconnect supervision. Another advantage of groundstart signaling is the ability for incoming calls (network -> CPE) to seize the outgoing channel, thereby preventing a glare situation from occurring. This is done by using the A- and B- bit on the network side instead of just the B-bit. The A-bit is also used on the CPE side. However, the B-bit can also be involved, based on the switch's implementation. Typically the B-bit is ignored by the Telco. This is a state table that defines this signaling information from the CPE's perspective (FXS).

This is the FXS-groundstart timing diagram.



On an incoming call (network-> CPE) this happens:

  1. The network goes off-hook and the A-bit goes from 1 to 0 and rings the line by toggling the B-bit between 0 and 1.
  2. CPE detects the ringing and seizure and goes off-hook and the A-bit is set to 1.
  3. The network goes off-hook and the B-bit stops toggling. B-bit is now 1.

In an outgoing call (CPE -> network) this happens:

  1. CPE goes ground on ring and A-bit and B-bit are 0.
  2. The network goes off-hook and the A-bit goes from 1 to 0. The B-bit is set to 1.
  3. The CPE goes off-hook. The A-bit and the B-bit are 1.
  4. CPE detects a dialtone and sends digits.

During a disconnect from the network, this occurs:

  1. The network goes on-hook and the A-bit goes from 0 to 1.
  2. CPE goes on-hook and the A-bit goes from 1 to 0.

During a disconnect from the CPE, the above steps are reversed.

EandM Signaling
E&M Signaling is typically used for trunk lines. The signaling paths are known as the E-lead and the M-lead. Descriptions such as Ear and Mouth were adopted to help field personnel determine the direction of a signal in a wire. E&M connections from routers to telephone switches or to PBXs are preferable to FXS/FXO connections because E&M provides better answer and disconnect supervision.

E&M signaling has many advantages over the previous CAS signaling methods discussed in this document. It provides both disconnect and answer supervision as well as glare avoidance. E&M signaling is simple to understand and is the preferred choice when you use CAS.

This is the E&M Signaling diagram.



The three types of E&M Signaling that are supported on Cisco routers are:

  • Wink-start (FGB) - Used to notify the remote side that it can send the DNIS information.
  • Wink-start with wink-acknowledge or double-wink (FGD) - A second wink that is sent to acknowledge the receipt of the DNIS information.
  • Immediate start - Does not send any winks at all.

Note: FGD is the only variant of T1 CAS that supports ANI and Cisco supports it along with the FGD-EANA variant. In addition to FGD functionality, FGD-EANA provides certain call services, such as emergency (USA-911) calls. With FGD, the gateway supports the collection of ANI inbound only. With the use of FGD-EANA, a Cisco 5300 is able to send ANI information outbound as well as collecting it inbound. This latter capability requires the user of the fgd-eana signaling type in the ds0-group command, with ani-dnis option and calling-number outbound command in the POTS dial-peer. The calling-number outbound command is supported only on the Cisco 5300 as of Cisco IOS Software Release 12.1(3)T.

Therefore, on an incoming call (network-> CPE) this process happens:

  1. The network goes off-hook. The A-bit and B-bit equal 1.
  2. CPE sends wink. The A-bit and B-bit equal 1 for 200 ms. This only occurs when you use wink-start or wink-start with wink acknowledgement. Ignore this step for immediate start.
  3. The network sends DNIS information. This is done by sending inband tones which are decoded by the modem.
  4. CPE sends a wink acknowledgement. A-bit and B-bit equal 1 for 200 ms. This only occurs for wink-start with wink acknowledgement. Ignore this step for immediate start or wink-start.
  5. CPE goes off-hook when a call is answered. A-bit and B-bit equal 1.

In an outgoing call (CPE -> network) the same procedure occurs. However, the network just described is the CPE and vice-versa. This is because the signaling is symmetric.

During a disconnect from the network, this process occurs:

  1. The network goes on-hook. A-bit and B-bit equal 0.
  2. CPE goes on-hook. A-bit and B-bit equal 0.

During a disconnect from the CPE, these two steps are reversed.

Telephony Application Programming Interface(TAPI)

由微軟 (MICROSOFT) 與英特爾 (INTEL) 所共同推動的 API 規格, 可使 WINDOWS 應用程式直接或透過網路 (NETWORK) 來控制電話相關設備, 例如數據機 (MODEM), 頭戴話機 (HEADSET), 交換機 (PBX) 等. TAPI 的目標是要建立一種標準的規格來控制從簡單的撥號到電話中心 (CALL CENTER) 的控制, 在 NETWARE 上運作類似的規格稱為 TSAPI(Telephony Server Application Programming Interface)。

TAPI 3.0 是集合傳統式 PSTN 電話服務和 IP 電話服務的漸進式 API。IP 電話服務是新方興的技術,能夠在現有的區域網路、廣域網路 和 Internet 上融合聲音、資料和視訊。 TAPI 3.0 讓IP 電話服務在 Microsoft® Windows® 作業系統上成為可能﹔這方法可以簡單而普通的方法結合二或多部電腦,並存取這種連接上所包納的所有媒體資料流。

TAPI 3.0 支援標準的 H.323 會議和 IP 多點傳送會議。它使用 Windows 2000 作業系統的 Active Directory 服務來簡化公司內的部署,包含服務品質 (QoS) 支援,以提高會議品質,使網路易於管理。


TAPI 3.0 提供了簡單而普通的方法,能夠結合兩部或多部電腦,並存取這種結合所涵蓋的任何媒體資料流。它摘錄了呼叫控制功能,讓不同而看似不相容的傳輸通訊協定提供應用程式使用的公用介面。h當公司開始從昂貴而缺乏彈性的電路交換公用電話網路轉移至有智慧、彈性而且便宜的 IP 網路時,IP 電話服務能夠自若地面對爆炸性的成長。TAPI 3.0 整合多媒體資料流控制和傳統電話服務。此外,它是從 TAPI 2.1 API 到 COM 模式的改進,可用任何語言編寫 TAPI 應用程式,例如 C/C++ 或 Microsoft® Visual Basic® 。

Java Telephony Application Programming Interface(JTAPI)

談到JTAPI(Java Telephony Application Programming Interface),首先得瞭解什麼是CTI。

CTI(Computer Telephony Integration)就是電腦電話集成技術,它是目前國內正火的呼叫中心熱潮的核心技術。JTAPI主要是為CTI技術服務。JTAPI(Java Telephone API)則是一套專門為JAVA語言提供的與電話應用相關的程式介面,它定義了一組跨平臺、跨廠家的電話應用程式物件模型。使用JTAPI提供的物件,我們就可以簡單方便地用軟體實現各種CTI技術。

由於JTAPI的誕生是由若干知名電腦、通訊廠商(Sun, Lucent Technologies, Nortel, Novell, Intel, and IBM)聯合努力的結果,利用JTAPI編寫的CTI程式甚至可以操作若干種電話交換機,這些交換機包括Lucnet、Nortel等等廠家。

JTAPI的主要特點歸納如下:
1. 簡化CTI程式的編寫。
2. 提供一套可以擴展的框架結構,可以平滑的使Client/Server結構的程式過渡到Browser/Server結構。
3. 對已有的傳統CTI程式介面,如TSAPI、SunXTL、以及TAPI進行WEB方向的擴展。
4. 可以運行於任何JAVA可以運行的平臺。

利用以上優點,採用JTAPI技術搭建的呼叫中心就可以平滑的過渡到Internet時代。

目前JTAPI主要應用於呼叫中心領域,利用它還可以編寫包括自動撥號、語音郵件、傳真接收等各類軟體。特別在互聯網呼叫中心領域更是大有用武之地。比如Lucent 推出的ICC(Internet Call Center)就是一個典型的例子。整個ICC系統從技術上劃分,可以分為3部分:管理、CTI、工作流。三個部分都用JAVA開發,其中CTI部分使用JTAPI1.3。利用JAVA的優勢,ICC可以運行在NT、SALORIS等各種平臺之上。

Cisco Survivable Remote Site Telephony (SRST)

Cisco SRST Description

Cisco SRST provides Cisco CallManager with fallback support for Cisco IP phones that are attached to a Cisco router on your local network. Cisco SRST enables routers to provide call-handling support for Cisco IP phones when they lose connection to remote primary, secondary, or tertiary Cisco CallManager installations or when the WAN connection is down.

Cisco CallManager supports Cisco IP phones at remote sites attached to Cisco multiservice routers across the WAN. Prior to Cisco SRST, when the WAN connection between a router and the Cisco CallManager failed or when connectivity with Cisco CallManager was lost for some reason, Cisco IP phones on the network became unusable for the duration of the failure. Cisco SRST overcomes this problem and ensures that the Cisco IP phones offer continuous (although minimal) service by providing call-handling support for Cisco IP phones directly from the Cisco SRST router. The system automatically detects a failure and uses Simple Network Auto Provisioning (SNAP) technology to autoconfigure the branch office router to provide call processing for Cisco IP phones that are registered with the router. When the WAN link or connection to the primary Cisco CallManager is restored, call handling reverts back to the primary Cisco CallManager.

When Cisco IP phones lose contact with primary, secondary, and tertiary Cisco CallManagers, they must establish a connection to a local Cisco SRST router to sustain the call-processing capability necessary to place and receive calls. The Cisco IP phone retains the IP address of the local Cisco SRST router as a default router in the Network Configuration area of the Settings menu. The Settings menu supports a maximum of five default router entries; however, Cisco CallManager accommodates a maximum of three entries. When a secondary Cisco CallManager is not available on the network, the local Cisco SRST router's IP address is retained as the standby connection for Cisco CallManager during normal operation.

Note: Cisco CallManager fallback mode telephone service is available only to those Cisco IP phones that are supported by a Cisco SRST router. Other Cisco IP phones on the network remain out of service until they reestablish a connection with their primary, secondary, or tertiary Cisco CallManager.

Typically, it takes three times the keepalive period for a phone to discover that its connection to Cisco CallManager has failed. The default keepalive period is 30 seconds. If the phone has an active standby connection established with a Cisco SRST router, the fallback process takes 10 to 20 seconds after connection with Cisco CallManager is lost. An active standby connection to a Cisco SRST router exists only if the phone has the location of a single Cisco CallManager in its CallManager list. Otherwise, the phone activates a standby connection to its secondary Cisco CallManager.

Note: The time it takes for an IP phone to fallback to the SRST router can vary depending on the phone type. Phones such as the Cisco 7902, Cisco 7905, and Cisco 7912 can take approximately 2.5 minutes to fallback to SRST mode.

If a Cisco IP phone has multiple Cisco CallManagers in its CallManager list, it progresses through its list of secondary and tertiary Cisco CallManagers before attempting to connect with its local Cisco SRST router. Therefore, the time that passes before the Cisco IP phone eventually establishes a connection with the Cisco SRST router increases with each attempt to contact to a Cisco CallManager. Assuming that each attempt to connect to a Cisco CallManager takes about one minute, the Cisco IP phone in question could remain offline for three minutes or more following a WAN link failure.

Note: During a WAN connection failure, when Cisco SRST is enabled, Cisco IP phones display a message informing you that they are operating in Cisco CallManager fallback mode. The Cisco IP Phone 7960G and Cisco IP Phone 7940G display a "CM Fallback Service Operating" message, and the Cisco IP Phone 7910 displays a "CM Fallback Service" message when operating in Cisco CallManager fallback mode. When the Cisco CallManager is restored, the message goes away and full Cisco IP phone functionality is restored.

While in Cisco CallManager fallback mode, Cisco IP phones periodically attempt to reestablish a connection with Cisco CallManager at the central office. Generally the default time that Cisco IP phones wait before attempting to reestablish a connection to a remote Cisco CallManager is 120 seconds. The time can be changed in Cisco CallManager; see the "Device Pool Configuration Settings" chapter in the Cisco CallManager Administration Guide. A manual reboot can immediately reconnect Cisco IP phones to Cisco CallManager.

Once a connection is reestablished with Cisco CallManager, Cisco IP phones automatically cancel their registration with the Cisco SRST router. However, if a WAN link is unstable, Cisco IP phones can bounce between Cisco CallManager and Cisco SRST. A Cisco IP phone cannot reestablish a connection with the primary Cisco CallManager at the central office if it is currently engaged in an active call.

Figure 1 shows a branch office with several Cisco IP phones connected to a Cisco SRST router. The router provides connections to both a WAN link and the PSTN. The Cisco IP phones connect to their primary Cisco CallManager at the central office via this WAN link.

Figure 1 Branch Office Cisco IP Phones Connected to a Remote Central Cisco CallManager


Figure 2 shows the same branch office telephone network with the WAN connection down. In this situation, the Cisco IP phones use the Cisco SRST router as a fallback for their primary Cisco CallManager. The branch office Cisco IP phones are connected to the PSTN through the Cisco SRST router and are able to make and receive off-net calls.

Figure 2 Branch Office Cisco IP Phones Operating in SRST Mode


H.323 Gateways and SRST
On H.323 gateways, when the WAN link fails, active calls from Cisco IP phones to the PSTN are not maintained by default. Call preservation may work with the no h225 timeout keepalive command, but call preservation using the no h225 timeout keepalive command is not officially supported by Cisco Technical Support.

Under default configuration, the H.323 gateway maintains a keepalive signal with Cisco CallManager and terminates H.323-to-PSTN calls if the keepalive signal fails, for example if the WAN link fails. To disable this behavior and help preserve existing calls from local IP phones, you can use the no h225 timeout keepalive command. Disabling the keepalive mechanism only affects calls that will be torn down as a result of the loss of the H.225 keepalive signal.

MGCP Gateways and SRST
MGCP fallback is a different feature than SRST and, when configured as an individual feature, can be used by a PSTN gateway. To use SRST as your fallback mode on an MGCP gateway, SRST and MGCP fallback must both be configured on the same gateway. MGCP and SRST have had the capability to be configured on the same gateway since Cisco IOS Release 12.2(11)T.

To make outbound calls while in SRST mode on your MGCP gateway, two fallback commands must be configured on the MGCP gateway. These two commands allow SRST to assume control over the voice port and over call processing on the MGCP gateway. The two commands are ccm-manager fallback-mgcp and call application alternate. A complete configuration for these commands is shown in the "Enabling SRST on an MGCP Gateway" section.

Note: The commands listed above are ineffective unless both commands are configured. For instance, your configuration will not work if you only configure the ccm-manager fallback-mgcp command.


Enabling SRST on an MGCP Gateway
To use SRST as your fallback mode with an MGCP gateway, SRST and MGCP fallback must both be configured on the same gateway. The configuration below allows SRST to assume control over the voice port and over call processing on the MGCP gateway.

Note: The commands described in the configuration below are ineffective unless both commands are configured. For instance, your configuration will not work if you only configure the ccm-manager fallback-mgcp command.

SUMMARY STEPS
1. enable
2. configure terminal
3. ccm-manager fallback-mgcp
4. call application alternate [application-name]
5. exit

Skinny Client Control Protocol(SCCP)

SCCP is a proprietary terminal control protocol originally developed by Selsius Corporation. It is now owned and defined by Cisco Systems, Inc. as a messaging set between a skinny client and the Cisco CallManager. Examples of skinny clients include the Cisco 7900 series of IP phone such as the Cisco 7960, Cisco 7940 and the 802.11b wireless Cisco 7920, along with Cisco Unity voicemail server. Skinny is a lightweight protocol which allows for efficient communication with Cisco CallManager. CallManager acts as a signaling proxy for call events initiated over other common protocols such as H.323, SIP, ISDN and/or MGCP.

A skinny client uses TCP/IP to and from one or more Call Managers in a cluster. RTP/UDP/IP is used to and from a similar skinny client or H.323 terminal for the bearer traffic (real-time audio stream). SCCP is a stimulus-based protocol and is designed as a communications protocol for hardware endpoints and other embedded systems, with significant CPU and memory constraints.

Cisco acquired SCCP technology when it acquired Selsius Corporation in the late 1990s. As a remnant of the Selsius origin of the current Cisco IP phones, the default device name format for registered Cisco phones with CallManager is SEP -- as in Selsius Ethernet Phone -- followed by the MAC address.

Other companies like Symbol Technologies and SocketIP have implemented this protocol in VoIP Terminals (phones) and Media Gateway Controllers or Softswitches. Open Source implementation of SCCTP/Skinny exist and are now part of the Asterisk (PBX) system.

A company named IPBlue has created a software phone (soft phone) which uses SCCP for signaling, too. This phone in fact appears to the Cisco CallManager server as a 7960 hardware phone.

In addition, Cisco has come out with its own version of a skinny softphone called Cisco IP Communicator as well as SIP-based softphone called Cisco Unified Personal Communicator. Previously, Cisco had a JTAPI/CTI version of a softphone called Cisco IP Softphone.

Impairment / Calculated Planning Impairment Factor (ICPIF)

The ICPIF originated in the 1996 version of ITU-T recommendation G.113 "Transmission impairments," as part of the formula Icpif = Itot - A. ICPIF is actually an acronym for "(Impairment) Calculated Planning Impairment Factor," but should be taken to simply mean the "calculated planning impairment factor." The ICPIF attempts to quantify, for comparison and planning purposes, the key impairments to voice quality that are encountered in the network.

ICPIF Stands for “Impairment Calculated Planning Impairment Factor”. The ICPIF attempts to quantify, for comparison and planning purposes, the key impairments to voice quality that are encountered in the network. ICPIF values are expressed in a typical range of 5(very low impairment) to 55 (very high impairment). ICPIF values numerically less than 20 are generally considered “adequate”

Note: IP SLA uses a simplified formula which is also used by Cisco Gateways to calculate the ICPIF for received VoIP data streams.

The ICPIF is the sum of measured impairment factors (total impairments, or Itot) minus a user-defined access Advantage Factor (A) that is intended to represent the user's expectations, based on how the call was placed (for example, a mobile call versus a land-line call). In its expanded form, the full formula is expressed as:

Icpif = Io + Iq + Idte + Idd + Ie - A

where

•Io represents impairments caused by non-optimal loudness rating,

•Iq represents impairments caused by PCM quantizing distortion,

•Idte represents impairments caused by talker echo,

•Idd represents impairments caused by one-way transmission times (one-way delay),

•Ie represents impairments caused by equipment effects, such as the type of codec used for the call and packet loss, and

•A represents an access Advantage Factor (also called the user Expectation Factor) that compensates for the fact that users may accept some degradation in quality in return for ease of access.

ICPIF values are expressed in a typical range of 5 (very low impairment) to 55 (very high impairment). ICPIF values numerically less than 20 are generally considered "adequate." While intended to be an objective measure of voice quality, the ICPIF value is also used to predict the subjective effect of combinations of impairments. Table 1, taken from G.113 (02/96), shows how sample ICPIF values are expected to correspond to subjective quality judgement.

Table 1 Quality Levels as a Function of Total Impairment Factor ICPIF

Upper Limit for ICPIF  Speech Communication Quality
5               Very good
10              Good
20              Adequate
30              Limiting case
45              Exceptional limiting case
55              Customers likely to react strongly
               (complaints, change of network operator)

For further details on the ICPIF, see the 1996 version of the G.113 specification.

Note:The latest version of the ITU-T G.113 Recommendation (2001), no longer includes the ICPIF model. Instead, it refers implementers to G.107: "The Impairment Factor method, used by the E-model of ITU-T G.107, is now recommended. The earlier method that used Quantization Distortion Units is no longer recommended."

The full E-Model (also called the ITU-T Transmission Rating Model), expressed as R = Ro - Is - Id - Ie + A, provides the potential for more accurate measurements of call quality by refining the definitions of impairment factors (see the 2003 version of the G.107 for details). Though the ICPIF shares terms for impairments with the E-Model, the two models should not be confused.

The IP SLAs VoIP UDP Operation feature takes advantage of observed correspondences between the ICPIF, transmission rating factor R, and MOS values, but does not yet support the E-Model.

Nov 28, 2007

MOS vs PSQM vs PAMS vs PESQ

清晰度與語音訊號可接受的程度有關,舉例來說,收訊者是否能聽得懂對方所說的話,由聲音辨別發話者是誰或是由聲音感受發話者的感覺。

由於清晰度的因果關係並非線性,因此在許多與語音壓縮有關的數位技術中(例如MPEG-2),清晰度會有所謂的臨界效應(cliff effect);所謂臨界效應是指隨著訊號損失的增加,清晰度會逐漸變差,當清晰度變差到一個程度之後,收訊者便完全無法聽清楚,"cliff"的實際位置通常得靠實驗決定。

傳統上,是以平均意見指標(mean opinion score, MOS)來衡量清晰度;平均意見指標是將收訊的語音樣本,由一群收訊者依收聽到的通話品質分成5個等級:1代表最差、5代表最佳,4則是一般公眾電話網路系統的通話品質。由於MOS很難建立一個客觀標準,而且有實際執行上的困難,因此MOS無法作為長期評估的標準。

為了改善MOS的這些缺點,陸續有人希望藉由電腦輔助的方式,提出各種具有重複客觀性通話品質的評量方法。大部分的方式都是由收訊者的觀點,來比較以人類自然語音訊號作為語音樣本經過傳輸之後,接受訊號和原始訊號之間的差異。

目前,常用的清晰度評量方法有兩種,一種是由荷蘭KPN Research所發展的知覺通話質量測量(Perceptual Speech Quality Measurement, PSQM),現已成為ITU-T P.861標準;另一種是由大英國協的英國電訊所發展的知覺分析/測量系統(Perceptual Analysis/Measurement System, PAMS)

PSQMPAMS都使用自然語音(natural speech)或類語音樣本作為輸入訊號,通常選擇的語音樣本(speech sample)會經由語音傳輸路經傳送,語音傳輸路經在經過編碼、封包化(packetization)、傳輸和解碼的過程中,會造成各種不同程度的訊號損失。評量的方法是以接收的語音樣本訊號,和原本的訊號作為清晰度演算法的輸入訊號。典型測試所採用的語音樣本會包括,具有各種代表性的男性和女性聲音。

PSQM演算法是以0到6.5的數字來評量清晰度,數字越低代表通話品質越好。PSQM原本是設計用來評估和比較各種語音編碼(speech codecs)技術的優劣,而非點對點的(end-to-end)網路通話品質。但是,加強許多功能之後(稱為PSQM+)便可用來作為網路通話品質測試,在比較PSQMMOS的時候必須特別注意,PSQM與傳統MOS聽音品質間的關係並非線性。

經驗顯示,如果系統可以提供更多的其他服務,使用者可接受比目前公眾電話網路略差的通話品質。

PAMS會產生聽音品質指標(listening quality score)(Ylq)和聽音效應指標(listening effort )(Yle)兩種指標,它們都是由0~15編排,數字越高代表品質越好。和PSQM清晰度指標一樣,聽音品質指標主要是評量收訊者接收的語音訊號,與原本訊號之間的相似度。至於聽音效應指標則是不同的評量方式。

聽音效應指標主要是針對嚴重失真無法以聲音品質評估的訊號,因此聽音效應指標評估的是,收訊者必須花費多少心力才能聽懂嚴重失真的語音訊號所傳遞的訊息。

至於評估PSQMPAMS這類客觀語音品質評量演算法是否有效的方法,則是比對PSQMPAMS指標與MOS測試結果間,是否具有明顯的相關性。通常這些客觀演算法與主觀MOS評量法之間的相關性高達r>0.9。至於其他傳統的客觀評量方法,如噪訊比(signal-to-noise ratio)與MOS之間的相關性則很差,所以即使噪訊比很高也無法保證具有良好的通話品質。

PSQMPAMS的開發者KPN Research與英國電訊最近共同合作提出新的客觀語音品質評量ITU-T標準,稱為語音質量感知評估(Perceptual Evaluation of Speech quality, PESQ)這項技術結合PSQMPAMS兩種方法的優點—PSQM的聽覺模型(perceptual model)和PAMS的時間對位法(time-alignment routine),所以PESQ指標與MOS指標g之間的相關性將更高。PESQ分數範圍從1(最差)到4.5(最好),3.8代表一般傳統付費電話的可接受語音品質。

聲音變化偵測器(Voice Activity Detector, VAD)

為了對頻寬做最有效的利用,我們必須偵測聲音的變化,並且視需要來啟動或是停止封包的傳輸。聲音變化偵 測器所必須解決的最大問題,就是如何分辨說話的內容以及伴隨而來的背景雜訊,我們可以利用這項功能來節省網路的頻寬,因為在一般的通話過程中,幾乎一半的時間都沒有人說話。

Electrical Echo vs Acoustic Echo

在語音基頻訊號(basic level)時,回音基本上是自己聽到自己的反射聲音;就技術上的觀點來看,如果訊號的接受路徑(receive path)出現傳輸訊號時,便會產生回音,最常見且希望出現的回音方式是側音(sidetone),亦即在電話筒中聽到自己沒有任何時間延遲的聲音。而實際上,如果講電話時聽不到側音時,大部分人都會懷疑自己講的話對方是否聽得見。

回音本質上可分為電訊回音(electrical echo)及聲音回音(acoustic echo)。電訊回音主要是由於串音(crosstalk)或是阻抗不匹配(poor impedance matching)所造成的,至於聲波回音最常見的情況,則為:喇叭與遠端的麥克風產生相互作用而產生聲波回音(Acoustic echo)。

回音所造成的訊號干擾,會隨著回音的強度和回音的時間延遲而不同[如圖1所示],由於側音的時間延遲很短,因此除非強度很大才會造成明顯的干擾。但是由於語音封包網路的延遲時間,通常比傳統語音網路的延遲時間長了10倍以上,所以干擾會變得較為明顯。

就原理上來說,處理回音的方法中,最直接和較簡單的便是抑制回音;抑制回音的方法是在傳輸訊號時,關閉收訊路徑(receive path)。但是這個方法的問題是回音抑制電路(echo suppressor circuitry)必須能偵測發話者結束發話的時間,所以會造成類似時間延遲過長,所形成的半工(half-duplex)通訊的問題。

由於抑制回音的方法會有上述問題,因此較佳的方式為抵銷回音,由於回音抵銷的高性能數位訊號處理器價格越來越便宜,所以目前逐漸改用抵銷回音的技術,來取代回音抑制技術。當回音的延遲時間很短時,抵銷回音可發揮最大效果;故此回音抵銷技術,通常會與其他減少系統時間延遲的技術一併使用。

回音抑制器(Echo Suppressor)

一種裝置將其插裝到網路中,使信號在網路中只能伸單方向之傳輸,反方向之信號被抑制或消除,如此可消除同被信號。在電話通信網路中,如兩個中繼站之間的距離超過1850哩時,就需要加裝回音抑制器,以保持通話聲音之清晰。

標準的數據描述語言ASN.1 (Abstract Syntax Notation One) 簡介

ASN.1是一種用描述結構化客體的結構和內容的語言.

抽象語法定義:
ASN.1是描述在網絡上傳輸信息格式的標準方法。它有兩部分:描述信息內數據,數據類型及序列格式的是一部分;另一部分描述如何將各部分組成消息。它原來是作為X.409的一部分而開發的,來才自己獨立成為一個標準。ASN.1在OSI的ISO 8824/ITU X.208(說明語法)和ISO 8825/ITU X.209(說明基本編碼規則)規范。下面就是一個例子:

Report ::= SEQUENCE {
author OCTET STRING,
title OCTET STRING,
body OCTET STRING,
biblio Bibliography
}

在這個例子中,"Report"是由名字類型的信息組成的,而SEQUENCE表示消息是許多數據單元構成的,前三個數據單元的類型是OCTET STRING,而最一個數據類型則下面的ASN.1語法表示它的意義:

Bibliography ::= SEQUENCE {
author OCTET STRING
title OCTET STRING
publisher OCTET STRING
year OCTET STRING
}

(http://www.fanqiang.com)

MGCP Endpoint Identifiers

在Cisco CVOICE 5.0 P.3-105最下方的Example: Endpoint Identifiers的例子中只有文字但是缺少了附錄圖片,可能會讓各位在研讀內容時摸不著頭緒,所以我在網路上找到了這一段原來的文章內容(我猜是舊版的教材只是不小心圖片的部份被刪掉了…),請參考以下內容:

When interacting with a gateway, the call agent directs its commands to the gateway for the express purpose of managing an endpoint or a group of endpoints. An endpoint identifier, as its name suggests, provides a name for an endpoint.

Endpoint identifiers consist of two parts: a local name of the endpoint in the context of the gateway and the domain name of the gateway itself. The two parts are separated by an at sign (@). If the local part represents a hierarchy, the subparts of the hierarchy are separated by a slash. In Figure 6-35, the local ID might be representative of a particular gateway/circuit #, and the circuit # might in turn be representative of a circuit ID/channel #. In Figure 6-35, mgcp.gateway.cisco.com is the domain name, and t1toSJ/17 refers to channel 17 in the T1 to San Jose.

Figure 6-35. MGCP Endpoint Identifier

WiMax網路 獲企業用戶青睞

【本報紐約訊】新一代無線寬頻傳輸技術WiMax趨於成熟,網路服務供應商紛紛利用這項高科技為企業建置網路環境,企業用戶可用更低廉的成本、享受更快速的上網品質,並掙脫長久以來電話公司幾近於獨占的強勢掌控。
根據美聯社報導,近年來,網路服務供應商(ISP,Internet Service Provider)善用無線傳輸技術挑戰電話公司幾近獨占的地位,但效果有限,直到最近WiMax技術成熟,網路服務供應商挾著這項高科技的威力再度出征,終於獲得企業界青睞。

總部位於麻州的塔流公司(Towerstream)是近幾年才創立的固定/無線寬頻服務供應商,正積極在全美各地推廣WiMax企業網路服務,營運觸角涵蓋紐約、邁阿密、洛杉磯、芝加哥、西雅圖、舊金山、波士頓等地;以紐約為例,目前在曼哈頓一棟27層樓高建築物屋頂架設天線,透過天線可為一定範圍內的用戶提供資料傳輸服務。

塔流公司執行長湯普森(Jeff Thompson)站在天線所在地眺望紐約市景,視線範圍從格林威治村一路延伸到曼哈頓中城,視線所及的每一棟大樓都是塔流公司的潛在客戶;該公司所提供的寬頻服務傳輸速度最高可達每秒10億位元(1 gigabit),一般狀況下,傳輸速率為每秒2000萬位元(20 megabit),消費端的下載速度與藉助其他技術的傳輸速度差別不大,但資料上傳速度快很多,對企業來說是一大突破。

史普林新一代(Sprint Nextel)也計畫今年底前在芝加哥開始提供WiMax寬頻網路傳輸服務;不過,執行長佛西(Gary Forsee)10月初被迫下台為這項計畫埋下變數,為求因應,已另外規劃與清晰線路公司(Clearwire)合作。以開發企業用戶為主的無線網路服務供應商也躍躍欲試,德州的iBroad-band便是一例。

塔流公司的運作模式是,向相關單位取得高聳建築物屋頂使用權、而後架設天線、在鄰近區域開發企業用戶,然而,這套模式並非萬靈丹,有些業者已宣告陣亡;上選公司(Best Buy)轉投資的Speakeasy2004年便以西雅圖地標建築物Space Needle為基地提供服務,卻無疾而終,塔流公司今年出面接收,打算起死回生。

塔流第三季核心業務毛利率高達59%,但由於斥資擴張設備,出現160萬元虧損、營收170萬元。

Newscast公司目前是塔流的客戶,之前曾向電話公司租用標準低頻網路線T1,這家企業的副總蘇萊表示,T1很不牢靠,而且費用高,改用塔流的服務後,每月只需要付500元,是T1月租的一半,而且傳輸速度是T1的三倍。

新一代無線寬頻傳輸技術WiMax趨於成熟,網路服務供應商紛紛利用這項高科技為企業建置網路環境;剛成立不久的塔流公司便在全美各地推廣WiMax企業網路服務。圖為塔流公司執行長湯普森在紐約市曼哈頓下城一棟高樓屋頂上,展示該公司興建的WiMax天線。【美聯社】

Nov 26, 2007

Configuring Echo Cancellation

Echo cancellation is configured at the voice port level. It is enabled by default, and its characteristics are configurable. Echo cancellation commands are as follows:
  • echo-cancel enable—Enables cancellation of voice that is sent out through the interface and received back on the same interface. Sound that is received back in this manner is perceived by the listener as echo. Echo cancellation keeps a certain-sized sample of the outbound voice and calculates what that same signal looks like when it returns as an echo. Echo cancellation then attenuates the inbound signal by that amount to cancel the echo signal. If you disable echo cancellation, it will cause the remote side of a connection to hear echo. Because echo cancellation is an invasive process that can minimally degrade voice quality, you should disable this command if it is not needed. There is no echo path for a four-wire E&RM interface. The echo canceller should be disabled for this interface type.
  • echo-cancel coverage—Adjusts the coverage size of the echo canceller. This command enables cancellation of voice that is sent out through the interface and received back on the same interface within the configured amount of time. If the local loop (the distance from the interface to the connected equipment that is producing the echo) is longer, the configured value of this command should be extended.

    If you configure a longer value for this command, it takes the echo canceller longer to converge. In this case, the user might hear a slight echo when the connection is initially set up. If the configured value for this command is too short, the user might hear some echo for the duration of the call because the echo canceller is not canceling the longer-delay echoes. There is no echo or echo cancellation on the network side (for example, the non-POTS side of the connection).
  • non-linear—The function enabled by the non-linear command is also known as residual echo suppression. This command effectively creates a half-duplex voice path. If voice is present on the inbound path, then there is no signal on the outbound path. This command is associated with the echo canceller operation. The echo-cancel enable command must be enabled for the non-linear command to take effect. Use the non-linear command to shut off any signal if near-end speech is not detected.

    Enabling the non-linear command normally improves performance. However, some users encounter truncation of consonants at the ends of sentences when this command is enabled. This occurs when one person is speaking and the other person starts to speak before the first person finishes. Because the nonlinear cancellation allows speech in one direction only, it must switch directions on the fly. This might clip the end of the sentence spoken by the first person or the beginning of the sentence spoken by the second person.

QSIG(Q SIGnaling)

QSIG(Q sinnaling)是一種數位綜合服務網的協定,在專用數位交換網基於Q931標準的一種傳輸信號。Q信號被廣泛使用在IP網、虛擬私人網路、高速多功能企業網、教育網和企業機關網。

Q信號在不同的銷售商生產的設備組成的結點間傳遞時確保了Q931的基本功能。這些功能包括啟動(表示連接建立的信號)、處理信號(說明信號被目的端處理的信號)、響鈴警告(告訴互交方目的結點正在響鈴)、連接(返回呼叫端說明目的節電已經接收到信號)、釋放/完成(說明發送方或接受放已經中止了信號)。Q信號分兩層:BC層(基本層)和CF層(產生層)。BC層用於遮罩硬體差別,使信號在結點間透明傳輸。Q信號GF層為大型企業網、教育網、政府機關網提供了附加功能,如線性鑒定、呼叫中斷、呼叫分發、多媒體應用等。

Configuring Hookflash Relay on FXS/FXO Voice Ports

Introduction
When you integrate Voice over IP (VoIP) technologies to legacy private branch exchange (PBX) and public switched telephone networks (PSTNs), there is sometimes a need to pass a type of signaling known as 'hookflash'. A hookflash is a brief interruption in the loop current on loopstart trunks that the attached system does not interpret as a call disconnect.

Once the PBX or PSTN senses the hookflash, it generally puts the current call on hold and provides a secondary dial tone or access to other features such as transfer or call waiting access.

A hookflash is done by momentarily pressing down the cradle on a telephone. Some telephone handsets have a button called 'flash' or 'recall' that sends a 'timed loop break', or 'calibrated flash' which is a hookflash that has a precise timing.

Background Information
Many customers use a combination of FXS and FXO ports to extend telephone handsets across IP networks. They want to preserve features of the existing PBX, such as call forward, no answer to voice mail, and transfer/hold on the remote extensions. Earlier Cisco VoIP software did not provide full control to allow transparent integration. However, with the release of H.323 version 2 support in Cisco IOS Software Release 12.0.5T and later, it is now possible to detect and pass hookflash signaling across IP networks.

When the FXS port is configured for a long 'hookflash in' timer value (greater than 500 msec), users may complain that when they hang up and immediately pick up the handset, the call has not cleared. If the value is set too low, the hookflash may be interpreted as a hang-up, but a higher value means the handset has to be left hung-up for a longer period to clear the call. In some cases, cradle bounce can cause problems as well. As the handset is hung-up, the spring tension on the hook button causes multiple short breaks on the line known as cradle bounce. Careful tuning of the hookflash in timing value may be needed for best results. One possibility in such cases is to use handsets with a flash button that sends a hookflash of a specific period. The FXO port can be set to match this value and the FXO port then generates the outgoing hookflash. Many PBXes have a Class of Service (CoS) option called 'calibrated flash' or 'timed loop break' which allows them to recognize hookflashes of specific duration and to ignore other shorter or longer loop breaks. Such settings are helpful in eliminating false disconnects and generation of invalid hookflash signals to the PBX.

Configure
In this section, you are presented with the information to configure the features described in this document.

Note: To find additional information on the commands used in this document, use the Command Lookup Tool ( registered customers only) .

Configure PLAR OPX and Hookflash Relay
Use this procedure to configure private line, automatic ringdown (PLAR) Off-Premises extension (OPX), and hookflash relay.

  1. Configure the FXO port on the MainSite router as connection plar-opx.

    The OPX mode allows remote users on FXS ports to appear to a central PBX as a directly connected extension. When the FXO port detects a ring signal from the PBX, the router sends a VoIP call setup to the remote FXS port but it does not take the FXO port off-hook. As a result, the PBX only sees the call answer signal when the RemoteSide router FXS port is picked up. After the PBX reaches the no answer timeout (call rings out), then it can end the call, transfer the call to voice mail, or ring another extension/ring group. Without OPX mode, the FXO port immediately goes off-hook after it senses the ringing and the PBX is then unable to perform a call forward, no answer, or roll over to voice mail.
  2. The RemoteSite router must be configured to sense and then pass the hookflash signal on the FXS port.

    Since the hookflash is a momentary break in the loop current on the FXS port and cannot be sent as an audio signal, the router passes the hookflash signal via dual tone multifrequency (DTMF) relay as the '!' character. The router with the FXO port then sends a short loop break which the external device sees as a hookflash. To properly pass the hookflash signal, the VoIP dial peers need to be configured for dtmf-relay h245-signal.
  3. The physical port timers have to be adjusted to suit the characteristics of the handset on the FXS port and the duration of the hookflash loop break out of the FXO port as shown here:

    。The FXS voice port (RemoteSite router) uses the timing hookflash-in msec command where msec is the maximum value of a loop break (in milliseconds) from the telephone handset that is interpreted as a hookflash. A loop break greater than the configured value is regarded as a disconnect and the call is dropped. Any interval under this value causes the router to send the '!' character via the H.245-signal DTMF relay.

    。The FXO voice port (MainSite router) uses the timing hookflash-out msec command where msec is the duration of the outgoing loop break in milliseconds. When the router receives an H.245-signal DTMF relay signal, the FXO port generates a loop break for the configured interval.

E&M Signalling Interface

Introduction
This appendix provides additional information on the tie line signalling standards and the FastPAD's E&M interface. The material presented here supplements the information provided in Chapter 7.

Signalling Types
The FastPAD supports five E&M signalling standards (Types I through V) for PBX tie line interfaces. These conventions, as defined by AT&T specifications, are described below.

With each signalling type, the PBX supplies one signal, known as the M signal (for Mouth), and accepts one signal, known as the E signal (for Ear). Conversely, the tie line equipment (e.g., the FastPAD) accepts the M signal from the PBX and provides the E signal to the PBX. The M signal accepted by the tie line equipment at one end of a tie circuit becomes the E signal output by the remote tie line interface.

Each of the five types is illustrated in Figure G-1. The illustrations in this figure are abstracted from the specifications to show the essential components of the signalling circuitry. In this Figure G-1, the symbol V refers to battery voltage, which can be 25 Vdc to 65 Vdc, and is usually (nominally) -48 Vdc. Each of the illustrations in the figure show the PBX's E&M interface on the left, and the corresponding tie line equipment interface on the right.

Type I
With the Type I interface the tie line equipment generates the E signal to the PBX by grounding the E lead. The PBX detects the E signal by sensing the increase in current through a resistive load (this is indicated in the Figure G-1 by the unconnected node branching from the right side of the E resistor). Similarly, the PBX generates the M signal by sourcing a current to the tie line equipment, which detects it via a resistive load.

The Type I interface requires that the PBX and tie line equipment share a common signalling ground reference. This can be achieved by connecting signal ground from the PBX to the SG lead (pin 8) of the RJ45 connector.

Type II
The Type II interface requires no common ground; instead, each of the two signals has its own return. For the E signal, the tie line equipment permits current to flow from the PBX; the current returns to the PBX's ground reference. Similarly, the PBX closes a path for current to generate the M signal to the tie line equipment.

Type III
A variation of Type II, Type III uses the SG lead to provide common ground. With this configuration, the PBX drops the M signal by grounding it, rather than by opening a current loop.

Type IV
Type IV is symmetric and requires no common ground. Each side closes a current loop to signal; the flow of current is detected via a resistive load to indicate the presence of the signal.

Type V
Type V is a simplified version of Type IV. This is also a symmetric interface, using only two wires. Type V requires a common ground between the PBX and the tie line equipment; this is provided via the SG leads.


Figure G-1: E&M Signalling Types



E&M Interface Types and Wiring Arrangement
There are five different E&M interface types or models named Type I, II, III, IV, and V (Type IV is not supported on Cisco platforms). Each type has a different wiring arrangement, hence a different approach to transmit E&M supervision signaling (on-hook / off-hook signaling). The signaling side sends its on-hook/off-hook signal over the E-lead. The trunking side sends the on-hook/off-hook over the M-lead.
For more information and pinout diagrams of E&M types, refer to Understanding and Troubleshooting Analog E&M Interface Types and Wiring Arrangements.

  • E&M Type I—This is the most common interface in North America.
    。Type I uses two leads for supervisor signaling: E, and M.
    。During inactivity, the E-lead is open and the M-lead is connected to the ground.
    。The PBX (that acts as trunk circuit side) connects the M-lead to the battery in order to indicate the off-hook condition.
    。The Cisco router/gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition.
  • E&M Type II—Two signaling nodes can be connected back-to-back.
    。Type II uses four leads for supervision signaling: E, M, SB, and SG.
    。During inactivity both the E-lead and M-lead are open.
    。The PBX (that acts as trunk circuit side) connects the M-lead to the signal battery (SB) lead connected to the battery of the signaling side in order to indicate the off-hook condition.
    。The Cisco router / gateway (signaling unit) connects the E-lead to the signal ground (SG) lead connected to the ground of the trunk circuit side in order to indicate the off-hook condition.
  • E&M Type III—This is not commonly used in modern systems.
    。Type III uses four leads for supervision signaling: E, M, SB, and SG.
    。During inactivity, the E-lead is open and the M-lead is set to the ground connected to the SG lead of the signaling side.
    。The PBX (that acts as trunk circuit side) disconnects the M-lead from the SG lead and connects it to the SB lead of the signaling side in order to indicate the off-hook condition.
    。The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate the off-hook condition.
  • E&M Type IV—This is not supported by Cisco routers / gateways.
  • E&M Type V—Type V is symmetrical and allows two signaling nodes to be connected back-to-back.
    。This is the most common interface type used outside of North America.
    。Type V uses two leads for supervisor signaling: E, and M.
    。During inactivity the E-lead and M-lead are open.
    。The PBX ( that acts as trunk circuit side) connects the M-lead to the ground in order to indicate the off-hook condition.
    。The Cisco router / gateway (signaling unit) connects the E-lead to the ground in order to indicate off-hook condition.

Application Examples
In examples below the term "attached device" refers to tie line equipment such as the FastPAD.

E&M Type I

Idle Condition. The E lead is biased by the PBX at -48 Vdc. The M lead is biased at 0 Vdc by the attached device (FastPAD).


PBX Initiated Call. The PBX supplies -48 Vdc to the M lead, signalling to the attached device that it wants a connection. The attached device grounds the E lead, signalling the response.


Attached Device Initiated Call. The attached device grounds the E lead, signalling to the PBX that it wants a connection. The PBX supplies -48 Vdc to the M lead, signalling the response.


PBX Initiated Disconnect. The PBX initiates disconnection by opening the M lead. The M lead is pulled to 0 Vdc, as biased by the attached device. In response, the attached device opens the E lead, which is pulled to -48 Vdc, as biased by the PBX.


Attached Device Initiated Disconnect. The attached device initiates the disconnect by opening the E lead. The E lead is pulled to -48 Vdc as biased by the PBX. In response, the PBX opens the M lead. The M lead, which is pulled to 0 Vdc, as biased by the attached device.

E&M Type II

Idle Condition. The E lead is biased by the PBX at -48 Vdc. The M lead is biased at -48 Vdc by the attached device (FastPAD).


PBX Initiated Call. The PBX grounds the M lead, signalling to the attached device that it wants a connection. The attached device grounds the E lead, signalling the response.


Attached Device Initiated Call. The attached device grounds the E lead, signalling to the PBX that it wants a connection. The PBX grounds the M lead, signalling the response.


PBX Initiated Disconnect. The PBX initiates disconnection by opening the M lead. The M lead is pulled to -48 Vdc, as biased by the attached device. In response, the attached device opens the E lead, which is pulled to -48 Vdc, as biased by the PBX.


Attached Device Initiated Disconnect. The attached device initiates the disconnect by opening the E lead. The E lead is pulled to -48 Vdc as biased by the PBX. In response, the PBX opens the M lead. The M lead, which is pulled to -48 Vdc, as biased by the attached device.

E&M Type V

Idle Condition. The E lead is biased by the PBX at -48 Vdc. The M lead is biased at -48 Vdc by the attached device (FastPAD).


PBX Initiated Call. The PBX grounds the M lead, signalling to the attached device that it wants a connection. The attached device grounds the E lead, signalling the response.


Attached Device Initiated Call. The attached device grounds the E lead, signalling to the PBX that it wants a connection. The PBX grounds the M lead, signalling the response.


PBX Initiated Disconnect. The PBX initiates disconnection by opening the M lead. The M lead is pulled to -48 Vdc, as biased by the attached device. In response, the attached device opens the E lead, which is pulled to -48 Vdc, as biased by the PBX.


Attached Device Initiated Disconnect. The attached device initiates the disconnect by opening the E lead. The E lead is pulled to -48 Vdc as biased by the PBX. In response, the PBX opens the M lead. The M lead, which is pulled to -48 Vdc, as biased by the attached device.

FastPAD E&M Interface
The FastPAD's E&M interface is designed to connect with that of a PBX tie line port, and provide appropriate end-to-end signalling support for a variety of applications. The following paragraphs describe this interface in detail.


FastPAD Circuits and E&M Signalling. The FastPAD generates the E signal to the PBX in response to an inbound signal at the remote FastPAD. That signal depends on the application of the remote unit. The local FastPAD will generate the E signal in the following applications:


The remote FastPAD is configured for E&M signalling, and detects an active M signal, The remote unit uses the FastPAD OPX option, and detects an off hook condition on its two-wire loop-start or ground-start circuit. The remote unit uses the FastPAD SLT option, and detects a ring signal on its two-wire loop-start or ground-start circuit.

Nov 24, 2007

Understanding and Troubleshooting Analog E&M Start Dial Supervision Signaling

Introduction
This document discusses analog recEive and transMit (E&M) Start Dial Supervision signaling. Start Dial Supervision is the line protocol that defines how the equipment seizes the E&M trunk and passes the address signaling information (sends dual tone multifrequency (DTMF) digits). The three main start dial supervision protocols used on E&M circuits are Immediate Start, Wink Start, and Delay Dial.

Wink Start Signaling
Wink is the most commonly used protocol. This is the Wink Start operation process:
1. Originating side seizes the trunk by going off-hook.
2. Terminating side remains idle (on-hook) until the digit collection equipment is attached.
3. Once the terminating side is ready, it sends a wink. A wink is an on-hook to off-hook to on-hook transition. This transition period ranges from 100 to 350 ms.
4. Once the origination side receives the wink, (which is interpreted as an indication to proceed), it sends the address (digits) information.
5. The call is then routed to its destination.
6. When the distant end answers, the terminating side signals answer supervision towards the originating side by going off-hook.
7. Both ends remain off-hook for the duration of the call.
8. Either end can disconnect the call by going on-hook.

The main reason for Wink Start (over Immediate Start) is to ensure that the side that receives the DTMF digits is ready to receive them. For PBX and central office (CO) products, the DTMF receivers are a shared resource and there may be less of them than there are total lines and trunks. Another reason is the glare reduction. Glare occurs when both ends of the trunk attempt to seize the trunk at the same time.



Immediate Start Signaling
Immediate Start signaling is the most basic protocol. The originating side goes off-hook, waits for a finite period of time (200 ms, for example), then sends the dial digits without regard to the far end (refer to the diagram).

The Immediate Start signaling method is less reliable than Wink Start. In Immediate Start, there is no wink from the end that receives the call to signify that it is ready to accept digits. In some situations, the PBX may be under heavy load and not able to switch a DTMF receiver in place quickly enough to receive the digits from the Cisco product. In that case, the call fails to complete because the Cisco product sends the DTMF digits before the PBX is ready to accept them. Therefore, for maximum reliability, Wink Start is preferred over Immediate Start.



Delay Dial Signaling
The Delay Dial operation process is shown here (refer to the diagram):

1. The originating side seizes the trunk by going off-hook.
2. The terminating side responds to the seizure by going off-hook.
3. The terminating side remains off-hook until it is ready to receive address information.
4. When the terminating side is ready, it goes on-hook. The off-hook interval is the delay dial signal.
5. The originating side starts sending address information.
6. The call is routed to its destination.
7. When the distant end answers, the terminating side signals answer supervision towards the originating side by going off-hook.
8. Both ends remain off-hook for the duration of the call.
9. Either end can disconnect the call by going on-hook.

Delay Dial is created because there are still problems in the field with Wink Start. There is equipment in the field that sends a wink, but it was not ready to receive digits the very instant after it sent the wink.

雙音多頻信號(Dual-Tone MultiFrequency, DTMF) vs 脈衝式撥號(Pulse Dialing)

雙音多頻信號(DTMF):

雙音多頻式電話,見下圖,顧名思義是建基於雙音調、多頻率 (Dual Tone Multi-Frequency, DTMF) 的概念。這種技術與傳統的十位脈衝式電話中以電脈衝的形式傳送訊號有所不同,DTMF電話所撥出的每個數字都由兩個音調組成,並以可聽得到的音調傳送到電話交換機。

雙音多頻式電話

DTMF電話配有一個按鍵式的撥號盤,上面有0至9的撥號數字,另外還有星號“*”及井號“#”用作完成一些特定的功能。如下圖所示,按鍵是以四列三行的二維陣列形式排列,對於每一列或行,都有特定頻率的音調。行的音調頻率較高而列的音調頻率則較低。當某一按鍵被按下時,由兩個不同頻率所組成的雙音調訊號會產生,這兩個頻率一個屬於低頻率群組而另一個則屬於高頻率群組。就是這個原因,所以我們才稱這種技術為“雙音多頻”。在這種技術中,由7個不同頻率的音調 (4個列頻率 + 3個行頻率) 可組成12種不同的頻率組合 (4 x 3)。舉例來說,若我們按下按鍵“5”,由770 Hz及1336 Hz組成的音調訊號會一起被傳送到電話交換機譯碼再分辨出所撥的是哪一個號碼。

按鍵號碼及其對應的頻率對應表


脈衝式撥號(PULSE DIALING):
使用於較老式的轉盤式號碼盤. 電話機,當號碼盤轉動時,以脈衝的多寡代表所播出的號碼。

迴路起動訊號(Loop start signaling) vs 接地起動訊號通知(Ground Start Signaling)

迴路起動訊號(Loop start signaling)

在家庭環境中,在本地電話總機房的電話交換機,可以根據電流是否流過本地迴路連線回到電話上來判別到底電話是被拿起聽筒(off-hook)還是被掛上聽筒(on-hook)的狀態。因為一隻聽筒被拿起來的電話就在機械方面來說它的尖塞(tip)和振鈴(ring)電路是開啟的,通過尖塞(tip)和振鈴(ring)所使用的-48伏特直流電並沒有做任何的事情。這個電壓只是在那邊等待著電路的關閉。當電話聽筒被拿起來之後,那麼,尖塞(tip)和振鈴(ring)電路關閉了,電流就可以開始流動通過這條電路。當電話總機房的電話交換器看見了這個電流開始流動,它便知道這隻電話已經被拿起聽筒,而且電話交換機會傳送撥號音(dial-tone)給這個發話者,告知他們可以開始撥打號碼了。這種類型的監督訊號傳送(supervisory signaling)被稱之為迴路起動訊號通知(loop start signaling)。

迴路起動訊號通知(Loop start signaling)有個可議之處就是睨視(glare)。您是否曾經拿起電話撥打給某人,但是您聽不到任何的撥號音卻發現某人正在電話線的另一端。如果是這樣子的情況,您就是發生了睨視(glare)。睨視(glare)又被稱之為通話碰撞(call collision),當同一個Trunk或是通道(channel)的兩端在同一個時間不約而同的拿起電話來準備建立通話時就會發生無法撥通的情況。

睨視(glare)在家庭環境中可能不是個大問題,但是對於連結到公司私人電話交換機(PBX)系統的線路會是怎麼樣呢?因為連接至私人電話交換機(PBX)的線路會發生明顯比您在家中電話使用更大量的通話,在PBX上使用迴路起動訊號發生睨視(glare)的可能性遠遠大於在您家中電話發生睨視(glare)的可能性。因此,您有時會發現在PBX以及在投幣式電話上使用了另一種類型的訊號傳送。也就是接地起動(ground start),這種訊號傳送的方式可以避免睨視(glare)現象的發生。

接地起動訊號通知(Ground Start Signaling)

利用接地起動訊號通知(ground start signaling),電話交換機監控著線路上電壓電位“振鈴(ring)”的提示,當振鈴(ring)提示擁有接地電位時,這條線路會被咬住。如果您開啟了電話聽筒的話筒部份,拉出其中一條導線(振鈴(ring)導線),並且讓那條導線接觸投幣電話的底盤(擁有接地電位)。透過建立這個拿起電話的地面起動訊號,那麼您就可以撥打電話了。

Nov 23, 2007

Lost Calls Cleared(LCC) vs Lost Calls Held(LCH) vs Lost Calls Delayed(LCD)

Lost Calls Cleared(LCC)(遺失通話清除)
LCC假設當電話被撥打時,伺服器(網路)正處於忙碌或是無法提供服務的狀態,該通話消失於系統中(沒有紀錄)。因此你放棄了並且嘗試去進行其他的動作。

Lost Calls Held(LCH)(遺失通話保留)
LCH假設一個電話在系統中維持保留的期間內,不論該電話號碼是否被撥打。因此在你放棄之前,你持續地去重撥直到保留時間結束。

Lost Calls Delayed(LCD)(遺失通話延遲)
LCD意謂當電話被撥打時,它維持在一個佇列中的等待時間直到伺服器準備好去處理這筆通話。

Note: LCC導致在主繼幹線(trunk)中的所需要的通話數量被低估;另一方面,LCH導致高估了在主繼幹線(trunk)中所需要的通話數量。

Poisson 分配

考慮下列現象:每小時服務台訪客的人數,每天家中電話的通數,一本書中每頁的錯字數,某條道路上每月發生車禍的次數,生產線上的疵品數,學生到辦公室找老師的次數……。大致上都有一些共同的特徵:在某時間區段內,平均會發生若干次「事件」,但是有時候很少,有時又異常地多,因此事件發生的次數是一個隨機變數,它所對應的機率函數稱為 Poisson 分配。


一個 Poisson 過程有三個基本特性:
(1) 在一個短時間區間 T 內,發生一次事件的機率與 T 成正比 。
(2) 在短時間內發生兩次以上的機率可以忽略。
(3) 在不重疊的時間段落裡,事件各自發生的次數是獨立的。

厄蘭(Erlangs)

一個公司的電話系統(例如,私人電話交換機(PBX))的通話數量利用厄蘭(Erlangs)來測量,厄蘭(Erlangs)是一小時的電話使用量。例如,如果您和您的三位同事每一個人使用您們的公司電話系統在相同小時中使用了30分鐘,那就是總共120通話分鐘(意即,4 * 30)的電話系統使用量。因為厄蘭(Erlangs)測量的單位是小時,您能將通話分鐘數除於60來轉變通話分鐘數成為厄蘭(Erlangs)。在本例中,您和您的同事擁有價值為2厄蘭(Erlangs)的電話系統使用量(意即,120 / 60)。

讓我們考慮一個方法來只需要您知道您的公司電話系統在一整個月期間內被使用的總分鐘數。這個“通話分鐘(call minutes)”數值可能來自於您公司的內部分機通話詳細紀錄或是來自於您公司的電話帳單。

據統計,一家公司電話系統在一天中忙碌小時(busy hour)期間內所經歷的通話分鐘數可以利用以下的公式來粗略估計︰

忙碌小時通話分鐘數(Busy_Hour_Call_Minutes) = [每月通話分鐘(Monthly_Call_Minutes) / 22] * 0.15

這個公式的基本原理基於觀察一個月包含大約22個營業日,並且在營業日中,一天通話量大約有百分之15發生在那天最繁忙小時期間。將忙碌小時通話分鐘數(Busy Hour Call Minutes)除於60,然後我們可以計算出來忙碌小時厄蘭(Erlangs)的數值。

舉例來說,假設有一家公司電話帳單顯示一個月的電話使用量是50,000分鐘。我們可以計算出來這家公司的忙碌小時通話分鐘數如下︰

忙碌小時通話分鐘數(Busy_Hour_Call_Minutes) = [50,000 / 22] * .15 = 340.9 通話分鐘

下一步,將通話分鐘數值除於60我們可以轉換通話分鐘數值變成厄蘭(Erlangs):

厄蘭(Erlangs) = 341 / 60 = 5.7

尖峰時間平均阻塞率(Grade of Service, GoS)

如果我們購買足夠的幹線(在私人電話交換機(PBX)世界中)或是足夠的頻寬(在網路電話(VoIP)世界中)來處理忙碌小時期間的每一個通話,某些幹線,或是頻寬,可能在一天的其餘時間裏並未被使用到。因此,我們必須確定一個在一天最忙碌小時期間內可接受的通話被拒絕的百分比。這個百分比被定義為尖峰時間平均阻塞率(grade of service,GoS)。

通常,電話網路設計者使用一個尖峰時間平均阻塞率(GoS)的百分比(P)(0.01),也就是一個通話在一天中最忙碌小時期間內被拒絕的機會是百分之一。

柔和噪音產生器(Comfort Noise Generator, CNG)

VoP是支援採用語音行為探測(VAD)的模式。在一般的電話通話過程中,50%或更多的時間都沒有語音信號,只有一些背景噪聲。如果在一個時間內沒有語音信號,可以不用傳送數據包。這會大幅降低對RTP處理器和網路呼叫客戶端的要求。在這種模式下運行時,通常感覺上語音品質比較差。但柔和噪音產生器(Comfort noise Generator)能藉由添加背景噪音來改善聲音效果,因為在話音間斷期間完全沒有聲音會讓人感覺不舒服。

因此,在進行網路閘道設計時,確定在何處產生RTP以及是否啟用VAD對設計工程師而言都非常重要,這些都會大幅影響VoP網路閘道設計的整體品質。

Nov 22, 2007

緊急回應定位(Emergency Response Location, ERL) vs 緊急定位辨識號碼(Emergency Location Identification Number, ELIN)

緊急回應定位(Emergency Response Location, ERL)指的是緊急電話(911/119)撥打的所在位置。在網路電話網路中常見VoIP號碼可攜,來電號碼並非總是代表緊急電話所在位置,因為發話者維持相同的代表號不論他們真實的所在位置。將設備及Port與一個ERL群組產生關聯性讓ERL可以被使用於號碼可攜網路中。

緊急定位辨識號碼(Emergency Location Identification Number, ELIN)是一個NANP(North American Numbering Plan)電話號碼用來使得緊急電話被導引至適當的PSAP。在一個可攜號碼網路中,來電號碼(Automatic Number Identification, ANI)是使用者所使用的可攜號碼,當通話被傳送至PSAP時,來電號碼就會被ELIN所取代。這樣的取代使得PSAP可以紀錄來電號碼並且可以回撥如果有這方面的需求時。每一個ERL都有一個唯一的ELIN相關聯。當一個漫遊的使用者登入網路電話系統時,這個網路電話號碼的ELIN就會根據網路電話連接的Port上相關聯的ERL而被確認。

公眾安全回應點 (Public Safety Answering Point, PSAP)

美國聯邦電信委員會(FCC)的強制要求(E911)

在1996年6月,美國聯邦電信委員會(FCC)正式通過一個報告書並要求擴大在無線911的服務(E911)。這個法規要求美國本土內的細胞式行動通訊系統(包括GSM,IS-95,cdma2000,...)、寬頻的個人通訊服務系統(PCS)等,要能傳遞撥話者的電話號碼至公眾安全回應點 (PSAP),並當有緊急電話(911)撥出能自動地轉給適當的PSAP,且即時地提供撥話者(發話手機)的所在位置。
無線的E911方案分成兩個階段,Phase I與Phase II。第一階段從2001年10月1日開始,其要求無線載波回報使用手機撥打911緊急電話的撥話者其電話號碼與撥話時的位置。第二階段要求無線載波提供更精準的位置資訊,一般為10米至100米。FCC針對第二階段制定一個4年的時程表,從2001年10月1日至2005年12月31日止。FCC針對 E911緊急情況服務命令的性能要求亦公佈一個適當的標準,依據使用定位技術的類別,針對性能上的要求亦有所不同。

緊急電話處理程序

1.用戶將話機使用地點通知Service Provider。
2.Service Provider將資料存入PSAP資料庫。
3.用戶撥打緊急電話(911/119)。
4.Call Server尋找處理緊急電話之PSAP。
5.緊急電話接通處理緊急電話之PSAP。
6.PSAP值班人員與用戶通話,有連接ALI(Automatic Location Identification)資料庫的PSAP能夠顯示發話者的電話號碼及發話位址資料。
7.PSAP通知相關緊急處理單位前往處理。

集中式數位交換機(CENTREX)系統

一般公司行號採購電話系統時,大部分都是自行購置私用交換機(PBX)。其實除了PBX之外,亦可選擇電信公司提供的CENTREX電話系統。 所謂CENTREX電話系統,中文翻譯為「集中式數位交換機」,簡稱CENTREX虛擬總機。就是在電信公司在市話交換機上,附加用戶專用交換機(PBX)軟體的功能,使在CENTREX系統之各線獨用電話可兼有私用交換機的分機功能。在CENTREX系統中,電信公司將市話交換機的部分用戶定義為一個基本用戶群,該用戶群的用戶不僅擁有普通用戶的所有功能,而且擁有私用交換機的所有功能。CENTREX系統用戶的分機可以獨立計費,也不會因中繼數量不足而產生話路壅塞;除此之外,若同一公司有多個辦公地點,而均在同一市話交換機之供線範圍內,則全部之分機皆可以視為同一套總機之分機,這是一般私用交換機所不及的地方。以前因為CENTREX線路的申租費用較高、大部分是學校等建築物分散的用戶在使用。但隨著固網業務開放,CENTREX線路租用費用日益降低,將吸引人更多企業用戶捨棄自行建置PBX而改用CENTREX系統。

CENTREX系統的功能

一般來說,電信公司為CENTREX用戶提供下列一般及特殊功能:

一.外線直接撥入分機(DID)
不必另外申請,外線就可以直接撥入任一分機,分機亦可以不經總機之轉接,直接抓外線撥號。

二.分機限撥
每一分機可以做適當之限制,如限撥外線、長途、國際或限外線直接撥入分機等之各項限制。

三.電話轉接功能
當電話進來,完成通話之後,如要再找第三人時,可以利用本功能,將電話轉接到第三人,繼續通話而不必重撥。

四.來話保留(Call Hold)
接到電話時,如需要轉問其他人員,而不希望對方知道通話內容時,可以利用本功能將對方保留起來,進行詢問動作;完成後,在接回對方,繼續服務工作。

五.仲介電話功能
控制者可以利用本功能,自行控制要與甲方或乙方通話,當與甲方通話時,乙方成保留狀態,而聽不到甲方與控制者之通話內容;反之亦同。

六.呼叫代接功能
同一單位之電話可以設定為同一帶接群,當有同事不再位置上,而有電話進來時,可以利用呼叫代接功能代接其電話,而不必到他的位置上幫他接電話。

七.條件指定轉接
當電話忙線與電話進來而無法應答時,將電話轉到所設定之電話。

八.自動回叫(Call Back)
當你有急事要與對方聯繫,而對方又在電話中,不知對方何時完成通話,就可啟動自動回叫之功能,由交換機協助監視對方,當對方完成通話,將電話掛掉時,交換機即刻像你振鈴,待你提起聽筒後,交換機在振鈴對方,達成你的通話需求。

九.話中插接(Call Waiting)
通話中,有電話進來時,交換機會用音響通知你有電話進來,你可以運用插撥功能在原通話者或是新進來之電話中切換通話對象。

十.無條件指定轉接(Call Forwarding)
本功能啟動後,當有電話進來時,就將電話轉到所設定之電話。

十一.三方通話:可以三人一起通電話。

十二.按時提醒(Wake Up Call)
類似鬧鈴功能;設定後,當時間到所設定之時間時,電話會以振鈴方式通知你。

十三.勿干擾功能
當有重要事情要處理,不希望被電話干擾時,啟動本項功能,交換機可以暫時幫你將電話擋掉,直到你將本功能取消為止。

十四.簡速撥號
你可以將常用之電話號碼,以代碼方式暫存到交換機內,就可以利用代碼方式來撥號,縮短撥號時間。

十五.熱線(Hot Line)
就是直通電話,可以將電話設定為熱線電話,一提起聽筒時,就將電話接預設之電話,達成快速通話之目的。



集中式數位交換機(CENTREX)與一般總機(PBX)之比較
一、集中式數位交換機(CENTREX)
1.裝設成本較低,客戶無需負擔折舊、維修、管理費用。且Centrex每一分機具有租用權,如不使用時,得過戶轉讓他人。
2.具直接撥入(DID),直接撥出(DOD)功能,可免設(或減少)值機員,節省人事費用支出。
3.話務疏通能力高,無中繼線不足問題。
4.具專用網路功能,不需考慮不同PBX間之匹配問題。
5.系統穩定性較PBX優。
6.無需裝機空間及電力空調等設備。
7.設備由中華電信提供、維修,客戶不需煩心。
8.同一市話交換局內,同機構內部分機間相互之通信免計次。

二、一般總機(PBX)
1.採購驗收手續麻煩,裝設成本較高,且需負擔折舊、管理、維修費用;分機無租用權。
2.需另備DID及DOD中繼器才可免設(或減少)值機員,人事費用支出較高。
3.中繼線不足,話務疏通能力較差。
4.需另租專線才能構成專用網路,尚須考慮不同PBX間之匹配問題。
5.系統穩定性遜於市話交換機。
6.需裝機空間及電力空調等設備。
7.PBX容量擴充或更新設備,須勞心勞力。
8.僅PBX分機裝設範圍內,其相互間通信免計次。

Nov 16, 2007

Gatekeeper-Routed Call Signaling(GKRCS) vs Direct Endpoint Signaling

There are two types of gatekeeper call signaling methods:

Direct Endpoint Signaling
—This method directs call setup messages to the terminating gateway or endpoint.

Gatekeeper-Routed Call Signaling (GKRCS)
This method directs the call setup messages through the gatekeeper.

Note: Cisco IOS gatekeepers are Direct Endpoint signaling based and do not support GKRCS.

These diagrams illustrate the differences between these two methods:



Back-to-back user agent(B2BUA)

From Wikipedia, the free encyclopedia

The Back-to-Back User Agent (B2BUA) acts as a user agent to both ends of a Session Initiation Protocol (SIP) call. The B2BUA is responsible for handling all SIP signalling between both ends of the call, from call establishment to termination. Each call is tracked from beginning to end, allowing the operators of the B2BUA to offer value-added features to the call.

To SIP clients, the B2BUA acts as a User Agent server on one side and as a User Agent client on the other (back-to-back) side. The basic implementation of a B2BUA is defined in RFC 3261. The B2BUA may provide the following functionalities:

  • call management (billing, automatic call disconnection, call transfer, etc.)
  • network interworking (perhaps with protocol adaptation)
  • hiding of network internals (private addresses, network topology, etc.)
  • codec translation between two call legs

Because it maintains call state for all SIP calls it handles, failure of a B2BUA affects all these calls. Often, B2BUAs also terminate and bridge the media streams to have full control over the whole session.

A Signaling gateway, part of a Session Border Controller, or Asterisk PBX are good examples of a B2BUA.

Dejitter

The dejitter buffer size determines the ability of the emulated circuit to tolerate network jitter. The dejitter buffer in CEoIP software is configurable up to 500 milliseconds; the maximum amount of network jitter that CEoIP can tolerate is ±250 milliseconds.

dejitter-buffer size
Example:
Router(config-cem)# dejitter-buffer 80

(Optional) Specifies the size of the dejitter buffer used to compensate for the network filter.
Use the size argument to specify the size of the buffer in milliseconds. Default is 60.

Quality of Service Options on GRE Tunnel Interfaces

The qos pre-classify command

When packets are encapsulated by tunnel or encryption headers, QoS features are unable to examine the original packet headers and correctly classify the packets. Packets traveling across the same tunnel have the same tunnel headers, so the packets are treated identically if the physical interface is congested. With the introduction of the Quality of Service for Virtual Private Networks (VPNs) feature, packets can now be classified before tunneling and encryption occur.

In the following example, tunnel0 is the tunnel name. The qos pre-classify command enables the QoS for VPNs feature on tunnel0:

Router(config)# interface tunnel0
Router(config-if)# qos pre-classify

Characterizing Traffic for QoS Policies

When configuring a service policy, you first may need to characterize the traffic that is traversing the tunnel. Cisco IOS supports Netflow and IP Cisco Express Forwarding (CEF) accounting on logical interfaces like tunnels. See the NetFlow Services Solutions Guide for more information.

Where Do I Apply the Service Policy?

You can apply a service policy to either the tunnel interface or to the underlying physical interface. The decision of where to apply the policy depends on the QoS objectives. It also depends on which header you need to use for classification.

  • Apply the policy to the tunnel interface without qos-preclassify when you want to classify packets based on the pre-tunnel header.
  • Apply the policy to the physical interface without qos-preclassify when you want to classify packets based on the post-tunnel header. In addition, apply the policy to the physical interface when you want to shape or police all traffic belonging to a tunnel, and the physical interface supports several tunnels.
  • Apply the policy to a physical interface and enable qos-preclassify when you want to classify packets based on the pre-tunnel header.

Cisco Security Device Manager(SDM) three categories

The Cisco SDM QoS wizard offers easy and effective optimization of LAN, WAN, and VPN bandwidth and application performance for different business needs (for example, voice and video, enterprise applications, and web). Three predefined categories are:

1. Real-time
2. Business-critical
3. Best-effort


In addition, the Cisco SDM QoS wizard supports NBAR, which provides real-time validation of application usage of WAN bandwidth against predefined service policies as well as QoS policing and traffic monitoring.

Nov 15, 2007

Police vs Shape

Policing can be applied to either the inbound or outbound direction, while shaping can be applied only in the outbound direction. Policing drops nonconforming traffic instead of queuing the traffic like shaping.

Policing also supports marking of traffic. Traffic policing is more efficient in terms of memory utilization than traffic shaping because no additional queuing of packets is needed.

Both traffic policing and shaping ensure that traffic does not exceed a bandwidth limit, but each mechanism has different impacts on the traffic:

1. Policing drops packets more often, generally causing more retransmissions of connection-oriented protocols, such as TCP.

2. Shaping adds variable delay to traffic, possibly causing jitter. Shaping queues excess traffic by holding packets in a shaping queue.

Traffic shaping is used to shape the outbound traffic flow when the outbound traffic rate is higher than a configured rate. Traffic shaping smoothes traffic by storing traffic above the configured rate in a shaping queue. Therefore, shaping increases buffer utilization on a router and causes unpredictable packet delays. Traffic shaping can also interact with a Frame Relay network, adapting to indications of Layer 2 congestion in the WAN. For example, if the backward explicit congestion notification (BECN) bit is received, the router can lower the rate limit to help reduce congestion in the Frame Relay network.

Processing vs Queuing vs Serialization vs Propagation vs End-to-End delay

1. Processing delay:
The time that it takes for a router (or Layer 3 switch) to take the packet from an input interface and put it into the output queue of the output interface. The processing delay depends on various factors:
  • CPU speed
  • CPU utilization
  • IP switching mode
  • Router architecture
  • Configured features on both the input and output interfaces

2. Queuing delay:
The time that a packet resides in the output queue of a router. Queuing delay depends on the number of packets already in the queue and their sizes. Queuing delay also depends on the bandwidth of the interface and the queuing mechanism.

3. Serialization delay:
The time that it takes to place a frame on the physical medium for
transport. This delay is typically inversely proportional to the link bandwidth.

4. Propagation delay:
The time that it takes for the packet to cross the link from one end to the other. This time usually depends on the type of media. (For example, satellite links produce the longest propagation delay because of the high altitudes of communications
satellites.)

5. End-to-end delay:
Equals the sum of all propagation, processing, serialization, and queuing delays in the path.

Convert Digital Signals to Analog Signals Steps

Step 1 Decompression:
If the voice signal was compressed by the sender, it is first decompressed.

Step 2 Decoding:
The received, binary formatted voice samples are decoded to the amplitude value of the samples. This information is used to rebuild a PAM signal of the original amplitude.

Step 3 Reconstruction of the analog signal:
The PAM signal is passed through a properly designed filter that reconstructs the original analog wave form from its digitally coded counterpart. The whole process is simply the reverse of the analog-to-digital conversion. Like analog-to-digital conversion, digital-to-analog conversion is performed by DSPs, which are located on the voice interface cards. The conversion is needed for calls being received from a packet network or digital interfaces, which are then transmitted out an analog voice interface.

FXS vs FXO vs E&M

Gateways use different types of interfaces to connect to analog devices, such as phones,
fax machines, or PBX or public switched telephone network (PSTN) switches. Analog
interfaces used at the gateways include these three types:

FXS:
The FXS interface connects to analog end systems, such as analog phones or
analog faxes, which on their side use the FXO interface. The router FXS interface
behaves like a PSTN or a PBX, serving phones, answering machines, or fax machines
with line power, ring voltage, and dial tones. If a PBX uses an FXO interface, it can also
connect to a router FXS interface. In this case, the PBX acts like a phone.

FXO:
The FXO interface connects to analog systems, such as a PSTN or a PBX, which
on their side use the FXS interface. The router FXO interface behaves like a phone,
getting line power, ring voltage, and dial tones from the other side. As mentioned, a PBX
can also use an FXO interface toward the router (which will then use an FXS interface),
if the PBX takes the role of the phone.

E&M:
The E&M interface provides signaling for analog trunks. Analog trunks
interconnect two PBX-style devices, such as any combination of a gateway (acting as a
PBX), a PBX, and a PSTN switch. E&M is often defined to as "ear and mouth," but it
derives from the term "earth and magneto." "Earth" represents the electrical ground, and
"magneto" represents the electromagnet used to generate tones.

Convert Analog Signals to Digital Signals Steps

Step 1 Sampling:
The analog signal is sampled periodically. The output of the sampling is a pulse amplitude modulation (PAM) signal.

Step 2 Quantization:
The PAM signal is matched to a segmented scale. This scale measures the amplitude (height) of the PAM signal.

Step 3 Encoding:
The matched scale value is represented in binary format.

Step 4 Compression:
Optionally, voice samples can be compressed to reduce bandwidth requirements. Analog-to-digital conversion is done by digital signal processors (DSPs), which are located on the voice interface cards. The conversion is needed for calls received on analog lines, which are then sent out to a packet network or to a digital voice interface.

Service-Provider CCIE Written 補充資料整理完結

經過了兩三個月的時間,我儘了最大的努力找出所有SP CCIE Written可能相關"主題"的文章摘要,並且用紅色字體標註"重點",希望對各位了解題意有幫助,順便可以知道原來的文章出處及更詳盡的內容意義。

今天已經完成了第一階段的SP CCIE Written考試,接下來就是要準備第二階段SP CCIE Lab的部份了,我會邊作Lab邊將個人心得紀錄在這個blog中,希望我的經歷可以讓各位更輕鬆地準備SP CCIE!

祝各位CCIE Candiates好運! 一起克服大魔王!

Why Are Some OSPF Routes in the Database but Not in the Routing Table?

Introduction
A common problem when using Open Shortest Path First (OSPF) is routes in the database don't appear in the routing table. In most cases OSPF finds a discrepancy in the database so it doesn't install the route in the routing table. Often, you can see the Adv Router is not-reachable message (which means that the router advertising the LSA is not reachable through OSPF) on top of the link-state advertisement (LSA) in the database when this problem occurs. Here is an example:

Adv Router is not-reachable
LS age: 418
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 172.16.32.2
Advertising Router: 172.16.32.2
LS Seq Number: 80000002
Checksum: 0xFA63
Length: 60

Number of Links: 3

There are several reasons for this problem, most of which deal with mis-configuration or a broken topology. When the configuration is corrected the OSPF database discrepancy goes away and the routes appear in the routing table. This document explains some of the more common reasons that can cause the discrepancy in the database.

Some of the commands used throughout this document for verification of OSPF behavior include the show ip ospf interface, ip ospf database router, show ip ospf neighbor and the show ip ospf database external . If you have the output of any of these commands from your Cisco device, you can use Output Interpreter to display potential issues and fixes. To use Output Interpreter , you must be a registered customer, be logged in, and have JavaScript enabled.

Reason 1: Network Type Mismatch
Let's use the following network diagram as an example:



R4-4K
interface Loopback0
ip address 172.16.33.1 255.255.255.255

interface Serial2
ip address 172.16.32.1 255.255.255.0
ip ospf network broadcast

router ospf 20
network 172.16.0.0 0.0.255.255 area 0

R1-7010
interface Loopback0
ip address 172.16.30.1 255.255.255.255
!
interface Serial1/0
ip address 172.16.32.2 255.255.255.0
clockrate 64000

router ospf 20
network 172.16.0.0 0.0.255.255 area 0

R4-4K(4)# show ip ospf interface serial 2
Serial2 is up, line protocol is up
Internet Address 172.16.32.1/24, Area 0
Process ID 20, Router ID 172.16.33.1, Network Type BROADCAST, Cost: 64
Transmit Delay is 1 sec, State DR, Priority 1
Designated Router (ID) 172.16.33.1, Interface address 172.16.32.1
Backup Designated router (ID) 172.16.32.2, Interface address 172.16.32.2
Timer intervals configured, Hello 10, Dead 40, Wait 40, Retransmit 5
Hello due in 00:00:08
Neighbor Count is 1, Adjacent neighbor count is 1
Adjacent with neighbor 172.16.32.2 (Backup Designated Router)
Suppress hello for 0 neighbor(s)

R1-7010(5)# show ip ospf interface serial 1/0
Serial1/0 is up, line protocol is up
Internet Address 172.16.32.2/24, Area 0
Process ID 20, Router ID 172.16.32.2, Network Type POINT_TO_POINT, Cost: 64
Transmit Delay is 1 sec, State POINT_TO_POINT,
Timer intervals configured, Hello 10, Dead 40, Wait 40, Retransmit 5
Hello due in 00:00:02
Neighbor Count is 1, Adjacent neighbor count is 1
Adjacent with neighbor 172.16.33.1
Suppress hello for 0 neighbor(s)
As you can see above, Router R4-4K is configured for broadcast, and Router R1-7010 is configured for point-to-point. This kind of network type mismatch makes the advertising router unreachable.

R4-4K(4)# show ip ospf database router 172.16.32.2

Adv Router is not-reachable
LS age: 418
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 172.16.32.2
Advertising Router: 172.16.32.2
LS Seq Number: 80000002
Checksum: 0xFA63
Length: 60
Number of Links: 3

Link connected to: another Router (point-to-point)
(Link ID) Neighboring Router ID: 172.16.33.1
(Link Data) Router Interface address: 172.16.32.2
Number of TOS metrics: 0
TOS 0 Metrics: 64

Link connected to: a Stub Network
(Link ID) Network/subnet number: 172.16.32.0
(Link Data) Network Mask: 255.255.255.0
Number of TOS metrics: 0
TOS 0 Metrics: 64

R1-7010(5)# show ip ospf database router 172.16.33.1

Adv Router is not-reachable
LS age: 357
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 172.16.33.1
Advertising Router: 172.16.33.1
LS Seq Number: 8000000A
Checksum: 0xD4AA
Length: 48
Number of Links: 2

Link connected to: a Transit Network
(Link ID) Designated Router address: 172.16.32.1
(Link Data) Router Interface address: 172.16.32.1
Number of TOS metrics: 0
TOS 0 Metrics: 64


You can see that for subnet 172.16.32.0/24, Router R1-7010 is generating a point-to-point link and Router R4-4K is generating a transit link. This creates a discrepancy in the link-state database, which means no routes are installed in the routing table.

R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 3 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
C 172.16.30.1/32 is directly connected, Loopback0


Solution
To solve this problem, configure both routers for the same network type. You can either change the network type of Router R1-7010 to broadcast, or change Router R4-4K's serial interface to point-to-point.

Note: If you have a situation where one side is a multipoint interface and the other side is a sub-interface then change the network type to broadcast on both sides.

In this example we have removed the "network-type broadcast" statement on R4-4K because both sides are point-to-point High-Level Data Link Control (HDLC) encapsulated interfaces.

R4-4K(4)# configure terminal
R4-4K(4)(config)# interface serial 2
R4-4K(4)(config-if)# no ip ospf network broadcast
R4-4K(4)(config-if)# end

R4-4K(4)# show ip ospf interface serial 2
Serial2 is up, line protocol is up
Internet Address 172.16.32.1/24, Area 0
Process ID 20, Router ID 172.16.33.1, Network Type POINT_TO_POINT, Cost: 64
Transmit Delay is 1 sec, State POINT_TO_POINT,
Timer intervals configured, Hello 10, Dead 40, Wait 40, Retransmit 5
Hello due in 00:00:04
Neighbor Count is 1, Adjacent neighbor count is 1
Adjacent with neighbor 172.16.32.2
Suppress hello for 0 neighbor(s)


Reason 2: Wrong Address Assignment in DUAL Serial Link Setup
Consider this network diagram as an example:



R4-4K
interface loopback 0
ip address 172.16.35.1 255.255.255.255

interface Serial2
ip address 172.16.29.1 255.255.255.0
!
interface Serial3
ip address 172.16.32.1 255.255.255.0

router ospf 20
network 172.16.0.0 0.0.255.255 area 0

R1-7010
interface loopback 0
ip address 172.16.30.1 255.255.255.255

interface Serial1/0
ip address 172.16.32.2 255.255.255.0
clockrate 64000
!
interface Serial1/1
ip address 172.16.29.2 255.255.255.0
clockrate 38400

router ospf 20
network 172.16.0.0 0.0.255.255 area 0


You can see that the IP addresses are flipped in the above configurations, which causes a discrepancy in the OSPF database. However, the routers still form neighbors in Cisco IOS version earlier than 12.1 because on a point-to-point link, OSPF routers don't verify that the neighboring router is on the same subnet.

R4-4K(4)# show ip ospf neighbor

Neighbor ID Pri State Dead Time Address Interface
172.16.32.2 1 FULL/ - 00:00:37 172.16.32.2 Serial2
172.16.32.2 1 FULL/ - 00:00:31 172.16.29.2 Serial3


From the above output, you can see that Serial2 is forming neighbors with IP address 172.16.32.2, which isn't in the same subnet. Although neighbors are formed, no routes are installed in the routing table:

R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 3 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
C 172.16.29.0/24 is directly connected, Serial1/1
C 172.16.30.1/32 is directly connected, Loopback0


Solution
To solve this problem, either correctly assign the IP addresses or switch the serial cables. Here we have corrected the IP addresses:

R4-4K
interface loopback 0
ip address 172.16.35.1 255.255.255.255

interface Serial2
ip address 172.16.32.1 255.255.255.0
!
interface Serial3
ip address 172.16.29.1 255.255.255.0

router ospf 20
network 172.16.0.0 0.0.255.255 area 0

R1-7010
interface loopback 0
ip address 172.16.30.1 255.255.255.255

interface Serial1/0
ip address 172.16.32.2 255.255.255.0
clockrate 64000
!
interface Serial1/1
ip address 172.16.29.2 255.255.255.0
clockrate 38400

router ospf 20
network 172.16.0.0 0.0.255.255 area 0


R4-4K(4)# show ip ospf neighbor

Neighbor ID Pri State Dead Time Address Interface
172.16.32.2 1 FULL/ - 00:00:36 172.16.32.2 Serial2
172.16.32.2 1 FULL/ - 00:00:39 172.16.29.2 Serial3


Now it shows the correct neighbor address on the Serial 2 interface. The routes are also in the routing table:

R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 4 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
O 172.16.35.1/32 [110/65] via 172.16.32.1, 00:03:12, Serial1/0
[110/65] via 172.16.29.1, 00:03:12, Serial1/1
C 172.16.29.0/24 is directly connected, Serial1/1
C 172.16.30.1/32 is directly connected, Loopback0

Reason 3: One Side of Point-to-Point Link Included in Wrong Majornet or Subnet
Consider this network diagram as an example:



This situation creates exactly the same behavior as the Wrong Address Assignment in DUAL Serial Link Setup. To solve the problem, assign IP addresses in the same subnet on both routers.

Reason 4: One Side Is Unnumbered and the Other Side Is Numbered
Consider the following network diagram as an example:



R4-4K
interface Loopback0
ip address 172.16.35.1 255.255.255.255

interface Serial2
ip unnumbered Loopback0
router ospf 20
network 172.16.0.0 0.0.255.255 area 0


R1-7010
interface Loopback0
ip address 172.16.30.1 255.255.255.255
!
interface Serial1/0
ip address 172.16.32.2 255.255.255.0
clockrate 64000

router ospf 20
network 172.16.0.0 0.0.255.255 area 0

R4-4K(4)# show interface serial 2
Serial2 is up, line protocol is up
Hardware is cxBus Serial
Interface is unnumbered. Using address of Loopback0 (172.16.35.1)

R1-7010(5)# show interface serial 1/0
Serial1/0 is up, line protocol is up
Hardware is cxBus Serial
Internet address is 172.16.32.2/24


The output above shows that R4-4K's Serial 2 interface is unnumbered to Loopback0, whereas R1-7010's Serial 1/0 is a numbered interface.

R4-4K(4)# show ip ospf interface serial 2
Serial2 is up, line protocol is up
Internet Address 0.0.0.0/24, Area 0
Process ID 20, Router ID 172.16.35.1, Network Type POINT_TO_POINT, Cost: 64
Transmit Delay is 1 sec, State POINT_TO_POINT,
Timer intervals configured, Hello 10, Dead 40, Wait 40, Retransmit 5
Hello due in 00:00:02
Neighbor Count is 1, Adjacent neighbor count is 1
Adjacent with neighbor 172.16.32.2
Suppress hello for 0 neighbor(s)


R1-7010(5)# show ip ospf interface serial 1/0
Serial1/0 is up, line protocol is up
Internet Address 172.16.32.2/24, Area 0
Process ID 20, Router ID 172.16.32.2, Network Type POINT_TO_POINT, Cost: 64
Transmit Delay is 1 sec, State POINT_TO_POINT,
Timer intervals configured, Hello 10, Dead 40, Wait 40, Retransmit 5
Hello due in 00:00:02
Neighbor Count is 1, Adjacent neighbor count is 1
Adjacent with neighbor 172.16.33.1

Suppress hello for 0 neighbor(s)

As you can see above, the network-type in both cases is point-to-point. The problem is that one side is unnumbered and the other side isn't, which creates a discrepancy in the database as shown below.

R4-4K(4)# show ip ospf database router 172.16.30.1

OSPF Router with ID (172.16.35.1) (Process ID 20)
Router Link States (Area 0)
LS age: 202
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 172.16.30.1
Advertising Router: 172.16.30.1
LS Seq Number: 80000002
Checksum: 0xC899
Length: 60
Number of Links: 3
Link connected to: another Router (point-to-point)
(Link ID) Neighboring Router ID: 172.16.35.1
(Link Data) Router Interface address: 172.16.32.2
Number of TOS metrics: 0
TOS 0 Metrics: 64
Link connected to: a Stub Network
(Link ID) Network/subnet number: 172.16.32.0
(Link Data) Network Mask: 255.255.255.0
Number of TOS metrics: 0
TOS 0 Metrics: 64
Link connected to: a Stub Network
(Link ID) Network/subnet number: 172.16.30.1
(Link Data) Network Mask: 255.255.255.255
Number of TOS metrics: 0
TOS 0 Metrics: 1

R4-4k(4)#

R1-7010(5)# show ip ospf database router 172.16.35.1

OSPF Router with ID (172.16.30.1) (Process ID 20)
Router Link States (Area 0)
Adv Router is not-reachable
LS age: 396
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 172.16.35.1
Advertising Router: 172.16.35.1
LS Seq Number: 80000003
Checksum: 0xBEA1
Length: 48
Number of Links: 2
Link connected to: another Router (point-to-point)
(Link ID) Neighboring Router ID: 172.16.30.1
(Link Data) Router Interface address: 0.0.0.3


!--- In case of an unnumbered link we use MIB
!--- II IfIndex value which usually starts with 0.


Number of TOS metrics: 0
TOS 0 Metrics: 64
Link connected to: a Stub Network
(Link ID) Network/subnet number: 172.16.35.1
(Link Data) Network Mask: 255.255.255.255
Number of TOS metrics: 0
TOS 0 Metrics: 1

R1-7010(5)#

You can see that R1-7010 is generating an LSA for this point-to-point link with the Link Data field containing its interface address, while R4-4K is generating the LSA for the same link with the Link Data field containing the MIBII IfIndex value. This creates a discrepancy in the link-state database, which means no routes are installed in the routing table.

R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 3 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
C 172.16.30.1/32 is directly connected, Loopback0


Solution
To solve this problem, configure both routers' serial interfaces as either numbered or unnumbered. In this example we have numbered the serial 2 interface of router R4-4K.

R4-4K(4)# configure terminal
R4-4K(4)(config)# interface serial 2
R4-4K(4)(config-if)# no ip unnumbered loopback 0
R4-4K(4)(config-if)# ip address 172.16.32.1 255.255.255.0

R4-4K(4))# show ip ospf interface serial 2
Serial2 is up, line protocol is up
Internet Address 172.16.32.1/24, Area 0
Process ID 20, Router ID 172.16.33.1, Network Type POINT_TO_POINT, Cost: 64
Transmit Delay is 1 sec, State POINT_TO_POINT,
Timer intervals configured, Hello 10, Dead 40, Wait 40, Retransmit 5
Hello due in 00:00:02
Neighbor Count is 1, Adjacent neighbor count is 1
Adjacent with neighbor 172.16.32.2
Suppress hello for 0 neighbor(s)

R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 3 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
O 172.16.33.1/32 [110/65] via 172.16.32.1, 00:03:08, Serial1/0
C 172.16.30.1/32 is directly connected, Loopback0

Reason 5: Broken PVC in Fully Meshed Frame Relay Environment
Consider this network diagram as an example:



R9-2500
interface Loopback0
ip address 50.50.50.50 255.255.255.255
!
interface Serial0
ip address 10.10.10.5 255.255.255.0
encapsulation frame-relay
ip ospf network broadcast
frame-relay map ip 10.10.10.6 102 broadcast
frame-relay map ip 10.10.10.7 101 broadcast

router ospf 10
network 10.10.10.0 0.0.0.255 area 0
network 50.50.50.0 0.0.0.255 area 0

R4-4K
interface Loopback0
ip address 70.70.70.70 255.255.255.255
!
interface Serial0
ip address 10.10.10.7 255.255.255.0
encapsulation frame-relay
ip ospf network broadcast
frame-relay map ip 10.10.10.5 101 broadcast
frame-relay map ip 10.10.10.6 100 broadcast

router ospf 10
network 10.10.10.0 0.0.0.255 area 0
network 70.70.70.0 0.0.0.255 area 0

R3-4K
interface Loopback0
ip address 60.60.60.60 255.255.255.255
!
interface Serial0
no ip address
encapsulation frame-relay
!
interface Serial0.1 multipoint
ip address 10.10.10.6 255.255.255.0
ip ospf network broadcast
frame-relay map ip 10.10.10.5 102 broadcast
frame-relay map ip 10.10.10.7 100 broadcast
!
router ospf 10
network 10.10.10.0 0.0.0.255 area 0
network 60.60.60.0 0.0.0.255 area 0


The broadcast model over Frame Relay works properly as long as the Frame Relay cloud is fully meshed. If any permanent virtual circuits (PVCs) are broken, it can create problems in the OSPF database, which in turn produces the Adv router not reachable message.

In this example, the PVC between R9-2500 and R4-4K is broken, and R9-2500 link to the designated router (DR) is broken. As a result, R9-2500 declares all LSAs from R3-4K (which is not a DR), as unreachable. As you can see, R9-2500 isn't generating a transit link for the serial interface attached to R3-4K; it is generating a stub link instead because as far as R9-2500 is concerned there is no DR on this link.

R9-2500(3)# show ip ospf database router

OSPF Router with ID (50.50.50.50) (Process ID 10)
Router Link States (Area 0)
LS age: 148
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 50.50.50.50
Advertising Router: 50.50.50.50
LS Seq Number: 8000000B
Checksum: 0x55A
Length: 48
Number of Links: 2

Link connected to: a Stub Network
(Link ID) Network/subnet number: 10.10.10.0
(Link Data) Network Mask: 255.255.255.0
Number of TOS metrics: 0
TOS 0 Metrics: 64

Link connected to: a Stub Network
(Link ID) Network/subnet number: 50.50.50.50
(Link Data) Network Mask: 255.255.255.255
Number of TOS metrics: 0
TOS 0 Metrics: 1

Adv Router is not-reachable
LS age: 1081
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 60.60.60.60
Advertising Router: 60.60.60.60
LS Seq Number: 80000006
Checksum: 0x4F72
Length: 48
Number of Links: 2

Link connected to: a Stub Network
(Link ID) Network/subnet number: 60.60.60.60
(Link Data) Network Mask: 255.255.255.255
Number of TOS metrics: 0
TOS 0 Metrics: 1

Link connected to: a Transit Network
(Link ID) Designated Router address: 10.10.10.7
(Link Data) Router Interface address: 10.10.10.6
Number of TOS metrics: 0
TOS 0 Metrics: 64


Adv Router is not-reachable
LS age: 306
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 70.70.70.70
Advertising Router: 70.70.70.70
LS Seq Number: 80000007
Checksum: 0xC185
Length: 48
Number of Links: 2

Link connected to: a Stub Network
(Link ID) Network/subnet number: 70.70.70.70
(Link Data) Network Mask: 255.255.255.255
Number of TOS metrics: 0
TOS 0 Metrics: 1

Link connected to: a Transit Network
(Link ID) Designated Router address: 10.10.10.7
(Link Data) Router Interface address: 10.10.10.7
Number of TOS metrics: 0
TOS 0 Metrics: 64


Refer to Problems with Running OSPF in NBMA Mode over Frame Relay for more detailed information about this problem.

Reason 6: Forwarding Address Known via an External Route
Consider this network diagram as an example:



R2507
interface Serial0
ip address 1.1.1.1 255.255.255.0

interface Serial1
ip address 7.7.7.1 255.255.255.0

router ospf 1
network 1.1.1.1 0.0.0.0 area 0
default- information originate metric 20

ip route 0.0.0.0 0.0.0.0 Serial1


R2504
interface Serial0
ip address 1.1.1.2 255.255.255.0

interface TokenRing0
ip address 3.3.4.2 255.255.255.0

router ospf 1
network 1.1.1.0 0.0.0.255 area 0
network 3.0.0.0 0.255.255.255 area 1
area 1 range 3.0.0.0 255.0.0.0


R2515
interface Serial1
ip address 4.4.4.3 255.255.255.0

interface TokenRing0
ip address 3.3.4.3 255.255.255.0

interface ethernet 0
ip address 3.44.66.3 255.255.255.0

interface ethernet 1
ip address 3.22.88.3 255.255.255.0

router ospf 1
redistribute rip metric 20 subnets
network 0.0.0.0 255.255.255.255 area 1

router rip
network 3.0.0.0


R2513
interface TokenRing0
ip address 3.3.4.4 255.255.255.0

interface ethernet 0
ip address 200.1.1.4 255.255.255.0

router rip
network 3.0.0.0
network 200.1.1.0

R2507# show ip ospf data external 200.1.1.0
OSPF Router with ID (7.7.7.1) (Process ID 1)
Type- 5 AS External Link States
LS age: 72
Options: (No TOS- capability, DC)
LS Type: AS External Link
Link State ID: 200.1.1.0 (External Network Number )
Advertising Router: 3.44.66.3
LS Seq Number: 80000001
Checksum: 0xF161
Length: 36
Network Mask: /24
Metric Type: 2 (Larger than any link state path)
TOS: 0
Metric: 20
Forward Address: 3.3.4.4
External Route Tag: 0


R2507 has 200.1.1.0/24 in its database but it hasn't installed it in the routing table because 3.3.4.4 is learned via an OSPF external route.

R2507# show ip route 3.3.4.4
Routing entry for 3.3.4.0/ 24
Known via "ospf 1", distance 110, metric 20,type extern 2, forward metric 70
Redistributing via ospf 1
Last update from 1.1.1.2 on Serial0, 00: 00: 40 ago
Routing Descriptor Blocks:
* 1.1.1.2, from 3.44.66.3, 00: 00: 40 ago, via Serial0
Route metric is 20, traffic share count is 1


Note: With the fix of Cisco bug ID CSCdp72526 ( registered customers only) , OSPF does not generate a type-5 link-state advertisement (LSA) of an overlapped external network; therefore, R2507 will only have a summary intra-area route of 3.0.0.0/8. Then, R2507 will install 200.1.1.0/24 as the forwarding address and it will be reachable via intra-area route 3.0.0.0/8, thus in compliance with RFC 2328 .

After the fix of above mentioned bug, output will look like the following:

R2507# show ip route 3.3.4.4
Routing entry for 3.0.0.0/8
Known via "ospf 1", distance 110, metric 74, type inter area
Last update from 1.1.1.2 on Serial0, 00:19:20 ago
Routing Descriptor Blocks:
* 1.1.1.2, from 3.3.4.2, 00:19:20 ago, via Serial0

R2507# show ip route
Codes: C - connected, S - static, R - RIP, M - mobile, B - BGP
D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area
N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2
E1 - OSPF external type 1, E2 - OSPF external type 2
i - IS-IS, su - IS-IS summary, L1 - IS-IS level-1, L2 - IS-IS level-2
ia - IS-IS inter area, * - candidate default, U - per-user static route
o - ODR, P - periodic downloaded static route

Gateway of last resort is not set

1.0.0.0/24 is subnetted, 1 subnets
C 1.1.1.0 is directly connected, Serial0
O IA 3.0.0.0/8 [110/74] via 1.1.1.2, 00:30:18, Serial0
O E2 200.1.1.0/24 [110/20] via 1.1.1.2, 00:22:58, Serial0
Route metric is 74, traffic share count is 1

R2507#

If the forwarding address is also known via an external route, OSPF doesn't install that route in the routing table. For more detailed information about this problem, see Common Routing Problem with OSPF Forwarding Address.

Reason 7: Distribute List Is Blocking the Routes
Let's use the following network diagram as an example:



R4-4K
interface Loopback0
ip address 172.16.33.1 255.255.255.255

interface Serial2
ip address 172.16.32.1 255.255.255.0

router ospf 20
network 172.16.0.0 0.0.255.255 area 0

R1-7010
interface Loopback0
ip address 172.16.30.1 255.255.255.255
!
interface Serial1/0
ip address 172.16.32.2 255.255.255.0
clockrate 64000

router ospf 20
network 172.16.0.0 0.0.255.255 area 0
distribute-list 1 in
!
access-list 1 permit 172.16.32.0. 0.0.0.255

As you can see above, R1-7010 has the distribute-list command configured and it's only allowing the 172.16.32.0/24 address range to be installed in the routing table. In link-state protocols you can not really filter an LSA with the distribute-list command. The LSA will still be in the database; however the LSA will not be installed in the routing table.

R1-7010(5)# show ip ospf database router 172.16.33.1

LS age: 357
Options: (No TOS-capability, DC)
LS Type: Router Links
Link State ID: 172.16.33.1
Advertising Router: 172.16.33.1
LS Seq Number: 8000000A
Checksum: 0xD4AA
Length: 48
Number of Links: 3

Link connected to: another Router (point-to-point)
(Link ID) Neighboring Router ID: 172.16.32.2
(Link Data) Router Interface address: 172.16.32.1
Number of TOS metrics: 0
TOS 0 Metrics: 64


The distribute-list configuration command on R1-7010 is filtering the 172.16.33.1/32 network from being installed in the routing table.

R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 3 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
C 172.16.30.1/32 is directly connected, Loopback0


Solution
To solve this problem, configure R1-7010 and allow 172.16.33.0/24 in the access control list (ACL) so this network gets installed in the routing table.

R1-7010(5)# configure terminal
R1-7010(5)(config)# access-list 1 permit 172.16.33.0 0.0.0.255
R1-7010(5)(config)# end

R1-7010(5)# show ip access-list 1
Standard IP access list 1
permit 172.16.32.0, wildcard bits 0.0.0.255
permit 172.16.33.0, wildcard bits 0.0.0.255


R1-7010(5)# show ip route
172.16.0.0/16 is variably subnetted, 3 subnets, 2 masks
C 172.16.32.0/24 is directly connected, Serial1/0
O 172.16.33.1/32 [110/65] via 172.16.32.1, 00:00:08, Serial1/0
C 172.16.30.1/32 is directly connected, Loopback0