Static Routing問題(qaet000提問於20090327)

Travis teacher：
我是你的學生，你發的CCNA Lab（20090303版本）
第四單元–第三小題設定R1使用static routing，將10.3.0.0/16的網段改由r3為唯一的下一跳，該怎麼做？謝謝

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Travis 回答 :
Hi 你好,
感謝你提出這樣一個問題, 這題我一時不察, 題目出錯了,
原本我的本意是希望同學在作 OSPF Routing Lab 練習時, 能觀察到兩條 "等值路由" (equal-cost load balance) 的現象, 然後再使用 Static Routing 指定其中一條 --- 但是好像我在 R1, R2 與 R3 互連的三條 serial 連線的 bandwidth 搞錯了, 所以觀察不出來等值路由的現象.

所以請你將題目作如下的更動:
更動一: 將 R1 與 R2 的連線 bandwidth 由 64K 改為 128K
更動二: 將 R2 與 R3 的連線 bandwidth 由 64K 改為 128K
更動一: 將 R3 與 R1 的連線 bandwidth 由 128K 改為 64K

經過以上的更動後, 此時請你啟動 OSPF Routing (注意: 各個端口的頻寬設定必須符合題目的要求), 如此你將可以在 R1 路由器的 routing table 內觀察到兩條 10.3.0.0/16 的等值路由, 如以下所示:
R1#show ip route
O 10.3.0.0/16 [110/1563] via 192.168.123.2, serial 0/0/0
[110/1563] via 192.168.123.9, serial 0/0/1

此時, 你可使用如下的 Static Routing 配置, 將 10.3.0.0/16 的下一跳指向 R3:
R1(config)#ip route 10.3.0.0 255.255.0.0 192.168.123.9

當你完成上述的 Static Routing 配置完後, 再觀察 R1 的 routing table, 則兩條OSPF等值路由的現象會消失, 反之則只會出現一條 Static route, 如以下所示:
R1#show ip route
S 10.3.0.0/16 [1/0] via 192.168.123.9
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不曉得你看懂了沒有, 如果沒有, 可直接在本文章下面再張貼你的意見.

另外, 感謝你這次的提問, 讓我找到了題目出錯的地方, 我會將這次更動併入到 CCNA Lab-20090327 的版本, Thanks.

/Travis 20090318 AM 0:24
Thanks for

PBX vs. Keyswitch

• A key system or key telephone system is a multiline telephone system typically used in small office environments. Key systems can be noted for their expandablity and having individual line selection buttons for each connected phone line.
• A private branch exchange (PBX) is a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company for many business or for the general public.
  1. PBX make connections among the internal telephones of a private organization - usually a business - and aslo connect them to the public switched telephone network (PSTN) via trunk lines. Because they incorporate telephones, fax machines, modems, and more, the general term "extension" is used to refer to any end point on the branch.
  2. PBXs are differentiated from "key systems" in that users of key systems manually select their own outgoing lines, while PABXs select the outgoing line automatically. Hybrid systems combine features of both.
  3. Initially, the primary advantage of PBXs was cost savings on internal phone calls: handling the circuit switching locally reduced charges for local phone services. As PBXs gained popularity, they started offering services that were not available in the operator network, such as hunt groups, call forwarding, and extension dialing. In the 1960x a simulated PBX known as Centrex provided similar features from the central telephone exchange.
  4. PBXs offer any other callig features and capabilities:
  Auto attendant 自動服務
  Auto dialing 自動撥號
  Automatic directory services 自動目錄服務 (when callers can be routed to a given employee bye keying or speaking the letters of the employee's name)
  Call accounting 通話統計
  Call forwarding on busy or absence 呼叫轉接

CCNA心得區

歡迎通過 CCNA 認証考試的同學, 請在本文章後面張貼意見, 來分享你的考試心得

PMBOK Guide Chapter 1

Chapter 1 - Introduction

• 1.1 Purpose of the PMBOK Guide
  The primary of the PMBOK Guide is to identify that subset of the Project Management Body of Knowledge that is generally recognized as good practice.
  
• 1.2 What is a Project
  A project is a temporary endeavor undertaken to create a unique product, service, or result.
  
• 1.3 What is Project Management?
  Project Management is the application of knowledge, skills, tools and techniques to project activities to meet project requirement.
  
• 1.4 The PMBOK Guide Structure
• 1.5 Areas of Expertise
• 1.6 Project Management Context

CCNA 問題區

如果對 CCNA 內容有任何問題, 請在本文章後面張貼意見

Cisco VoIP Structure

The OSI reference model was designed to create standard methods of connecting and communicating across data networks. In the realm of voice, similar connection models exist to describe voice communications. Cisco has developed a type of model to describe the Unified Communication System.

Cisco Voice Unified Communications Layers

Endpoints Layer	IP Phone, Wireless/Cell phone, Video Phone, IM Client
Applications Layer	Voice Mail, Conference Call Apps, Call Center Apps, 911 Services
Call Processing Layer	Unified Communications Manager, Unified Communications Manager Express, UC500
Infrastructure Layer	ASA Firewall, Voice Router/Gateway, Voice Switch

While the layers work together to create a functional voice network, each layer performs unique functions to the voice network.

Common Channel Signaling (CCS)

Common channel signaling dedicates one of the DS0 channels from a T1 or E1 link for signaling information. This is often called “out of band” signaling because the signaling traffic is sent completely separate from the voice traffic. As a result, a T1 connection using CCS has only 23 usable DS0 for voice.

Because CCS dedicates a full channel of the circuit for signaling, the “stolen bit” method of signaling using ABCD bits is no longer necessary. Rather, a full signaling protocol sends the necessary information for all voice channels. The most popular signaling protocol used is Q.931, which is the signaling protocol used for ISDN circuits.

CCS is the most popular connection used between voice systems worldwide because it offers more flexibility with signaling messages, more bandwidth for the voice bearer channels, and higher security (because the signaling is not embedded in the voice channel). CCS also allows PBX vendors to communicate proprietary messages (and features) between their PBX systems using ISDN signaling, whereas CAS does not offer any of these capabilities.

Note: When using CCS configurations with T1 lines, the 24th time slot is always the signaling channel. When using CCS configurations with E1 lines, the 17th time slot is always the signaling channel.

Channel Associated Signaling (CAS)

T1 CAS

Because T1 CAS steals bits from the voice channel to transfer signaling information, it is often called robbed bit signaling (RBS).

The 24 channels of the digital T1 circuit carry only voice data for the first five frames that they send. On the sixth frame (marked with an S), the eighth bit (also called the least significant bit) is stolen for the voice devices to transmit signaling information. This process occurs for every sixth frame after this (12th, 18th, 24th, and so on). The stolen bit relays the signaling information for each respective DS0 channel. For example, the bits stolen from the third DS0 channel relay the signaling information only for that channel.

Notice that at the end of each frame sent for the 24 DS0 channels is an F, signifyinng the T1 framing bit. When a T1 digital line sends voice data, it does so by sending all 24 of the smaller DS0 frames in one big T1 frame. So each T1 frame is 193 bits in length. Here’s the math:

Each DS0 frame = 8 bits T1 sends 24 DS0 frame at once (8 bits * 24 frames) = 192 bits Each T1 frame has a framing bit (1 bit + 192 bit) = 193 bits Digital T1 lines send 8000 of these 193-bit frames every second (because Dr. Nyquist’s voice model requires 8000 samples to be sent each second to accurately reconstruct voice), THis is why T1 lines runs at 1.544Mbps: (193 bits per frame * 8000 frames per second) = 1,544,000 bps or 1.544 Mbps

Super Frame (SF) sends groups of 12 T1 frames at a time. When using SF, all 12 of the T1 framing bits are used to keep the T1 equipment synchronized with the other side. This means that all 8000 T1 framing bits sent every second are dedicated to synchronization.

The newer standard, Extended Super Frame (ESF), sends groups of 24 T1 frames at a time. The newer ESF standard uses the framing bit more intelligently than the older SF standard. Of the 8000 bits every second, ESF is able to use 2000 bits for synchronization, 2000 bits for error checking, and 4000 bits as supervisor channels, which is able to send control functions and perform error reporting.

Note: All modern T1 service providers use ESF.

E1 CAS

E1 lines have 32 channels, which break down as follows:

• E1 DS0 1: Used for E1 framing information
• E1 DS0 2-16: Dedicated use for voice (no signaling)
• E1 DS0 17: Used for voice signaling information for channels 2-16 and 18-32
• E1 DS0 18-32: Dedicated use for voice (no signaling)

Note: It might seem odd to have channel associated sighnaling that is sent in a separate channel (channel 17) in E1 deployments, but that is how E1 CAS operates. It is still considered CAS because the signaling sent in time solt 17 uses the same system of ABCD bits as T1 CAS.

Problems of analog connections

Two original problems of analog connections:
從類比訊號的問題導入數位訊號來解決

• The signal degrades over long distances.
  - Digitizing voice solves the first problem because you can easily transmit a numeric value any distance a cable can run without any degradation or line noise.
  
• You cannot send multiple calls over a single line.
  - Time-Division multiplexing (TDM) solves the second problem. TDM allows voice networks to carry multiple conversations at the same time over a single, four-wire path. Because the multiple conversations have been digitized, the numeric values are transmitted in specific time slots (thus the “time-division”) that differentiate the separate conversations.
  

A T1 circuit is built from 24 separate 64-kbps channels known as a digital signal 0 (DS0). Each one of these channels is able to support a single voice call. A E1 circuit allows you to use up to 30 DS0s for voice calls.

Although digital technology solves the problems of signal degradation and the inability to send multiple calls over a single line that occur in analog technology, it creates a new issue: signaling (這裡的signaling指的是Supervisory/Informational/Address signaling). To solve this, two primary style of signaling were created for digital circuits:

• Channel associated signaling (CAS): Signaling information is transmitted using the same bandwidth as the voice.
  
• Common channel signaling (CCS): Signaling information is transmitted using a separate, didicated signaling channel.

DTMF vs. Pulse

Analog Connections - Address Signaling

• Dual-tone multifrequency (DTMF): The buttons on a telephone keypad use a pair of high and low electrical frequencies (thus "dual-tone") to generate a signal each time a caller presses a digit.
  
• Pulse: The rotary-dial wheel of a phone connects and disconnects the local loop circuit as it rotates around to signal specific digits.

Loop Start vs. Ground Start Signaling

Analog Connections - Loop Start and Ground Start Signaling

Loop Start Signaling:

• When the phone is lifted off-hook, the phone connects tthe two wires, causing an electrical signal (48V DC voltage) to flow from the phone company central office (CO) into the phone. This is known as loop start signaling.


• Disadvantage:
  Glare problem occurs when you pick up the phone to make an outgoing call at the same time as a call comes in on the phone line before the phone has a chance to ring.


• Typically used home environments.
  

Ground Start Signaling:

• The grounding of the wires would signal the phone company to send a dial tone on the line. Using this type of signaling in PBX systems allows the PBX to separate an answering phone from an incoming phone line, reducing the problem of glares.


• Because of glare, most modern PBX systems designed for larger, corporate environments use ground start signaling.

Loop start vs. Ground Start Signaling

In order to receive a dial tone from the CO, the PBX must send a ground signal on the wires. This intentionally signals to the telephone CO that an outgoing call is going to happen, whereas using the loop start method of signaling just connects the wires to receive an incoming call or place an outgoing call.