Learning IOS and JUNOS

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.72...

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 bit...

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® 。

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 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 Simp...

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...

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....