NCNU VoIP Course

Acer Aspire M1641

2008-10-19T09:44:00.000+08:00

To install a new server, I spent $9,990 to buy a desktop PC.
It is Acer Aspire M1641 E2200. I also spent $500 to buy extra 1GB DDRII-800 memory.
However, it is not easy to install Windows XP on this model.

The first problem I encounter is that WinXP cannot detect the SATA-320 hard drive.
You need to prepare a floppy disk which contains the driver of SATA hard drive.
Yet, there is a problem - this model does not include a floppy drive, so you need to
find a floppy drive to attach to this PC before it can install Windows XP to its hard drive.
(This process requires you to open the case, but that is all right.)

Because I don't have a spare floppy drive, the engineer in Sinya helps me to do this.
He installed Windows XP on H: drive. Well, this is not perfect, but acceptable.

Now here comes the second problem.
I use the Acer Driver CD-ROM to install the drivers of display card and network interface card,
but failed in installing the High Definition Audio card. I downloaded driver on Acer's website,
but the version is the same as the one in the CD-ROM (ver. 1.8). I'll ask the engineer to check it
for me again.

Can we have an IVR system?

2007-12-18T17:12:00.000+08:00

今年二月台南市曾經辦理一個200人的小規模的網路電話說故事活動

http://summer.tn.edu.tw/telstory96/

當時因為達聲沒辦法提供技術支援，所以最後是以e-mail人工作業方式來處理留言的語音檔，程序是

1.學生先到網頁上報名，報名完成後才會知道先撥06-7000818再轉66666即可留言。

2.學生留言完成後，語音檔會自動e-mail寄給活動管理者，管理者再依照語音開頭唸的班級姓名分類。

3.進行評審。

比較好的流程應該是

1.學生先到網頁上報名，報名完成後才會取得留言號碼及密碼。

2.學生利用網路電話撥66666或家中電話撥06-7000818再轉66666，即可進入網路電話說故事IVR平台。

3.接著學生輸入留言號碼及密碼，選擇聽取錄音或開始錄音，最長錄音時間五分鐘。

4.錄製完成後，IVR平台自動將語音檔對應到網頁連結，學生也可以從網頁登錄後聽取錄音。

5.活動管理者可選擇做篩選或直接公佈錄音。

想來想去，其實問題很明確：請問有沒有符合上述需求的IVR平台可以使用？

如果有IVR平台且能以號碼xxxxx註冊到台南市的達聲sip server，似乎還可進行一些很有趣的活動，如

許願樹：新年新希望

主題討論：發表對某一人事物的看法

交作業：老師指派朗誦或背誦作業，請學生完成

TANet VoIP交換中心

2007-12-08T20:53:00.000+08:00

上次詢問的TANet VoIP 交換中心運作事宜，聯繫後確認這個案子由瑪凱得標，不過負責交換的NextTone設備由金誠科技維護，相關聯絡人如下：

話機裝機問題:

林慶南
金誠科技股份有限公司
02-8712-7080#826
0918-141-322

互聯問題:

王鵬龍 ( Long Wang )

金誠科技
mobile phone - 0953-810635
office phone - 02-87127080 ext 822

昨天建議金誠科技應該要準備好SOP和表格，讓要加入交換的小管理區有標準流程可以依循，希望他們能儘速改善。

教育部管理區的設備由瑪凱維護，據說還沒有整合好，所以還無法加入TANet VoIP交換中心。

沒有自己sip server的縣市網，由這次標案提供一台顯赫的sip server(163.28.2.1)，每個縣市網各有1台video+5台ip phone註冊到顯赫的sip server。

以台南市網為例，現行的作法是

---------------------------------

教育部端設定(交換用NextTone163.28.2.7, 話機註冊用顯赫163.28.2.1)
163.28.2.7信任163.26.2.3
Prefix 9211000往163.28.2.1送
Prefix 921往163.26.2.3
收到來自台南市來電，將原來的5碼再加上921，改以921xxxxx送出。

台南市端設定(台南市話機註冊用163.26.2.3)
163.26.2.3信任163.28.2.7
Prefix 9211000往163.28.2.7送
收到921xxxxx來電，去掉921後送到163.26.2.3
Prefix 9往163.28.2.7送
---------------------------------

921台南市網現在已經可以和以下二個單位互通

918新竹市網

988高雄縣網(Loxa網路電話)

有空時可以聯繫一下金誠科技，將暨大也加入TANet VoIP交換

Cannot get IPv4 address from DHCP server

2006-12-12T12:52:00.000+08:00

Dear sir,
Sorry! I have fixed the problem.
Reason:
We use 10.10.20.0 (secondary) and 163.22.20.0 in vlan 20. Dhcp server 's
setting
should have both setting, but only do not delivery 163.22.20.xxx . My
colleage took away
163.22.20.0 setting. Router transfers client's request by using
163.22.20.254, but at last
it found no match.
Way to fix:
1. add 163.22.20.xxx in config file. (I afraid it will take long time,
and I have no rights)
2. change 163.22.20.0 to secondary and 10.10.20.0 be the primary. (I
have done it)

leeh

Quincy Wu 提到:

>救命呀! 我現在整個房間都沒辦法從 DHCP Server 拿到 IPv4 address
>(但 IPv6 拿得到).
>
>如果設定成 public IP address, 就一切正常運作 (還好之前多申請了幾個).
>麻煩檢查一下 DHCP server.
>
>Sincerely,
>Quincy Wu
>Nov. 28
>
>.
>
> 
>

Apache HTTPD 2.0

2006-09-10T12:48:00.000+08:00

I am trying to install Apache HTTPD 2.0 with SSL support, so that we may manage the mailinst list of "VoIP over WiMAX" project through a secure transport protocol.

Server version: Apache/2.2.0
apachectl startssl - this option has been obsoleted. To start httpd with SSL support, you should edit your configuration file to include the relevant directives and then use the normal apachectl start.
vi /usr/local/etc/apache22/httpd.conf
apachectl configtest
SSLCertificateFile: file '/usr/local/etc/apache22/server.crt' does not exist or is empty
vi /etc/ssl/openssl.cnf

countryName_default = TW
stateOrProvinceName_default = Nantou
localityName_default = Puli
0.organizationName_default = National Chi Nan University
organizationalUnitName_default = Graduate Institute of Communication Engineerin
g
commonName_default = ns3.ipv6.club.tw

/usr/bin/openssl genrsa -out /usr/local/etc/apache22/server.key 1024
/usr/bin/openssl req -new -days 365 -key /usr/local/etc/apache22/server.key -out /usr/local/etc/apache22/server.csr
/usr/bin/openssl x509 -in /usr/local/etc/apache22/server.csr -out /usr/local/etc/apache22/server.crt -req -signkey /usr/local/etc/apache22/server.key -days 365
chmod 400 server.*

Now https is working!

[Price] Hard Phone

2005-07-12T16:10:00.000+08:00

莊組長：

新竹市網路電話話機已於昨日決標，得標公司『金葉科技』、話機為FCI代工之話機，決標單價：每支2029元。總共：580支。

預計7/31交貨。

拔銓

Polycom 視訊會議設備

2005-07-11T11:34:00.000+08:00

暨南大學吳坤熹博士提到有關於VSX7000連線到VSX3000視訊主機問題部份：

1.VSX7000_VC至VSX3000無法顯示Data畫面，回應VSX3000軟體更新須從7.5.1版升級至7.5.2版。

2.SIP 多點，目前無法做到，7.5.2版後即更新為8.0版，屆時便可以支援SIP多點。

3.為何VSX選項裏有防火牆設定選單，因為就其認知，無論使用NAT或Mapping動作，皆由防火牆設定即可，為何視訊主機仍有防火牆設定選單，其功能為何？

4.關於Qos設定問題，已回應OK，係有5個選項，預設為最優先。

國科會新修訂「補助專題研究計畫經費處理原則」

2005-06-27T10:18:00.000+08:00

通知 94年6月23日

一、國科會新修訂「補助專題研究計畫經費處理原則」，請轉知，並請依說明事項辦理專題研究計畫經費支用等相關事宜。

二、業務費部分：修正將「研究人力費」、「耗材物品及雜項費用」、「國合計畫國外學者來台費用」合併為「業務費」，此三細項經費如有不敷使用或剩餘，可逕行於業務費項下調整。

三、自94年8月1日起，「研究人力費」之專任助理人員工作酬金依執行機構訂定之標準支給，本項請依本組另案通知辦理。

四、研究設備費部分：修訂因執行計畫需要必須變更設備費項目時，如擬變更之單價超過(含)30萬 (原為10萬) 元以上，應報經國科會同意後始可變更及採購；單價未達30萬元之變更，則校內程序簽准即可辦理。

五、國外差旅費部分：將「國外或大陸地區差旅費」、「出席國際學術會議差旅費」、「國合計畫出國差旅費」合併為「國外差旅費」，此三項費用之出國人員、人數、次數、天數或地點變更，須經校內程序簽准後於原核定經費下列支。

六、另配合【科學技術基本法】第六條增修「公立學校…不適用政府採購法之規定」，本次修訂「耗材物品及雜項費用」、「研究設備費」進行採購時應依執行機構內部審核規定及政府補助科學技術研究發展採購監督管理辦法之規定辦理。相關作業疑義請洽購運組及會計室。

七、檢附「國科會補助專題研究計畫經費處理原則簡易對照表」，此表僅供參考用，其他注意事項仍請詳參經費處理原則。配合此次修訂，本校內部簽核使用之「國科會計畫經費變更處理表」業已修正，學校權限以內之項目變更，請逕至本組網頁國科會資訊下載最新表格填送申請。

交大研發處計畫業務組啟

分機：31580

[Book] Putting VoIP to Work

2005-06-22T11:17:00.000+08:00

Title: Putting VoIP to Work - Softswitch Network Design and Testing
Author: Bill Douskalis
Publisher: prentice Hall PTR, Upper Saddle River, NJ 07458, 2002.
ISBN: 0-13-040959-6

Archived in NCNU library.

Contents:

Protocols and Topologies of the New PSTN
Topologies, Equipment, and Communications
SONET and DWDM Technology Overview: The Present and the Future
The Pursuit of QoS in Packet Network
H.323: The Next Generation
Media Gateway Control Protocol
H.248: Megaco Protocol
SIP: The Evolution of the Session Initiation Protocol
PSTN Signaling Adaptation and Transport

Comments: I only read Chapter 7. It has more detailed explanation about the packages of MEGACO, in comparison with Daniel's "Carrier Grade VoIP".

Draft Agenda of RADVISION Training

2005-06-21T15:18:00.000+08:00

Instructor : Rachel Danor, support manager (From Radvision),
Amway Tsai, FAE manager & Thiessen Lee, FAE engineer (from E-Soft)
Date : July, 12(Tue.) - July 13(Wed.)
Location : NCTU campus (assigned by NTCU)
Training Agenda :
July, 12 (Tue.)
9:30-12:00 Radvision SIP Toolkit introduction (Rachel, speak in English)
13:00-17:00 Radvision SIP Toolkit Architecture and API programming (Rachel, speak in English)
July, 13 (Wed.)
9:30-12:00 Advance RTP/RTCP introduction (Rachel, speak in English)
13:00-15:00 ProLab Testing Tools introduction and hand-on (Amway and Thiessen speak in Mandarin)
15:30-17:00 Hand on demo (Amway and Thiessen speak in Mandarin)

[Book] PBX Systems for IP Telephony

2005-03-28T18:34:00.000+08:00

I saw a book in NCNU library: "PBX Systems for IP Telephony", written by Allan Sulkin in 2002.
(TK5105.8865 S949 2002) Its Appendix A lists some call processing features, such as

Station User Features

Add-on Conference
This feature allows a station user to add a third party to an existing two-party conversation.

Automatic Callback
This feature is used when a dialed station is busy. When the feature is activated, the system automatically attempts to callt he desired station until the line is free. The calling party is alerted that the called party is available. This saves wasted time dialing when encountering busy signals.

Automatic Intercom
This feature provides a talking path between two voice terminal users. A station user presses a programmed automatic intercom button and lifts the handset, or vice versa. The called user receives a unique intercom alerting signal, and the status lamp associated with the dial or automatic intercom button, if provided, flashes

Bridged Call Appearance
This feature allows the same line appearance to be programmed to appear on more than one telephone. It is very useful in manager-assistant relationships and call answering position environments. A bridged call appearance can reduce the number of abandoned or lost calls and allows the coverage station to prescreen calls for the called party.

Call-Back Last Internal Caller
This feature allows a station user to automatically consult and call back the last internal caller to the station (unanswered call) by implemententing the feature code.

Call Forwarding - All Calls/No Answer/Busy
This feature allows a station user to divert all incoming calls to another programmed station. All Call activation diverts all incoming calls to the station; no answer activation diverts calls after a programmed number of rings; busy activation diverts incoming calls when the station is busy. The features are useful when a station user is away from the desk area, wishes to receive calls at another station, or when there is a desire not to receive calls, but the user wants the call answered. The most common coverage station is a voice messaging port. Call forwarding features decrease abandoned or lost calls and improve call coverage service for the calling party.

Call Forwarding - Follow Me
This feature allows a station user to activate the call forwarding feature from a remote telephone by chainging an existing forwarding destination. It provides station users with the capability of changing call forwarding destinations without returning to their desks, and can be used to "follow" station users around the system if they wish to receive calls at different stations.

Call Forwarding - Off Premises
This feature allows a station to forward all calls to an off-premises location outside the system. It allows station users to receive calls at a programmed outside telephone line when they are out of the office.

Call Forwarding - Ringing
This feature allows a station user of a multiple line voice terminal with display to forward an incoming call during the ringing period to another station. The station user can read the display for screening information, such as CLID or calling party name, before activating the feature through a programmed feature button or dial access code. The rerouted call destination is input by the station user after feature activation. The operation is transparent to the calling party.

Call Forwarding - Selective Multiple Line
This feature allows a station user of a multiple line voice terminal to selectively call forward any or all line appearance numbers.

Call Hold
This feature allows a station user to place an existing call in a hold state when there is another incoming call or the station user must leave the desktop area for more than a few seconds. Call hold provides station users with the flexibility of handling multiple concurrent calls without re-establishing the connection after finishing another call.

Call Park
This feature allows a station user to "park" a call at the received station, effectively placing the call in a hold state, retrieve the call at another station, and contnue the conversation. A second party can also retrieve the parked call if notified by the first party. The feature provides station users with mobility and eliminates the need to return calls.

Call Pickup
This feature allows a station user to retrieve and answer a call directed to another station (direct), any station in the station user's assigned call pickup group (group), or another call pickup group (designated group). The station user presses a programmed call pickup button or dials the desired feature access code to implement the feature.

Call Transfer
This feature allows a station user to divert an existing call to another station within the system. It eliminates the need for the original calling party to hang up and redial another telephone line to reach the desired or proper called party.

Call Waiting
This feature notifies a station user engaged in conversation that there is another incoming call to the station line number. The notification is usually a special tone or display signal on the telephone. Call waiting reduces lost calls, improves customer service, and reduces the number of calls forwarded to another station or messaging system.

Consecutive Speed Dialing
For speed dialing all station number digits are registered as the speed dial code. This feature allows a common set of partial station number digits to be registered as the speed dial code, and allows the station user to dial the remaining digits of each number to establish the call.

Consultation (Broker) Hold
This feature allows a station user to place one party on hold and confer with a third party on another line. This feature reduces call backs and improves customer service. Some systems allow the station user to toggle back and forth between two lines.

Customer Station Rearrangement
This feature allows station users to physically relocate their multiple line voice terminals internal to the system. When a station user moves between locations, the voice terminal station (number and COS) is logically transferred. This service is activated as follows: dialing the "moving" feature code followed by a personal code; the terminal is out of service; the terminal is disconnected, moved, and reconnedted int he new location; the terminal is reinitialized by dialing its extension number, followed by the personal code; the station rings immediately and when answered, the set is validated. The former and latter locations must support the same category of voice terminal.

Discrete Call Observing
This feature allows a supervisor to monitor a conversation between an assistant and a caller on a preselected line. While the supervisor is listening, the voice terminal microphone is off and the assistant is informed that the feature is activated by a notice on the voice terminal display. During monitoring, the supervisor can take over the call.

Distinctive Ringing
This feature allows the system administrator to define distinct ringing patterns for different call types, such as internal, defined internal line, external, private network, emergency, and private line. Station users can use this feature as a call screening device to decide which calls to answer and which are to be forwarded to a coverage station. The number of distinct patterns differs greatly between different system models.

Dial by Name
This feature allows a station user with a voice terminal equipped with an alphabetical keyboard to call an internal extension or external number by typing in a name using last name, first name, or initials. The directory database can be locally stored in the voice terminal or accessed from a centrally located database in the PBX system or application.

Do Not Distrub
This feature allows a voice terminal user to request that no calls, other than priority calls, terminate at a particular extension number until a specified time. At the specified time, the system automatically deactivates the feature and allows calls to terminate normally at the extension.

Elapsed Call Timer
This feature provides a display of the elapsed time when a multiple line voice terminal is connected to any trunk circuit.

Emergency Access to Attendant
This feature allows emergency calls to be placed to an attendant with special priority status. Calls can be placed automatically when the telephone is in an off-hook state or by dial access. The attendant received visual and audible feedback when the call is received. The feature is important for situations requiring immediate attendant access.

Executive Busy Override
This feature allows a station user to cut into an internal party's conversation. This feature decreases call backs, saves time reaching the called party, and decreases calls sent to coverage positions.

Executive Calling
This feature allows a station to be assigned VIP class status. The feature allows a VIP station to send a special ringing signal to a called station when idle and automatically send multiple tone bursts to that station when busy.

Excecutive Access Override
This feature allows a station user to connect a call to an internal extension that is in call forward or do not disturb mode. The service is authorized by COS level.

External Paging with Meet-Me
This feature allows a station user or attendant to dial a local paging equipment access code and connect both parties automatically after the paged party has answered the page and dials a special access code.

Facility Busy Indication
This feature provides multiple line telephone users with a visual indication of the busy or idle status of internal station numbers, trunk groups, hunt groups, or paging zones. Station users can monitor the activity of frequently called numbers with this feature, eliminating encountered busy signals.

Group Listening
This feature allows a station user of a multiple line voice terminal with an integrated speakerphone to place a call using the handset and activate the terminal's built-in speaker, to allow others to listen to the conversation while the station user continues talking through the handset.

Hands-free Dialing
This feature allows a station user of a voice terminal with a built-in speaker to dial and monitor a call without lifting the handset.

Hands-free Intercom
This feature allows a station user of a voice terminal with a built-in speaker to answer a voice call without lifting the handset. The incoming voice call is ehard over the speaker.

Help/Information Key
This feature provides a station user of a multiple line voice terminal with display immediate access to help menus for terminal programming and feature access procedures. Information is displayed by the system in user-friendly way. If a feature access code is changed, it will be displayed automatically in the feature menu. Service consultation shows all the relevant functions and their associated feature codes. This feature allows self-traning on the voice terminal and also reduces the need for paper labels on telephone features.

Hot Line
This feature automatically dials calls to preassigned internal stations, off-premises stations, or feature access codes when the handset is lifted. It eliminates the need to dial a number or access code, thereby simplifying and accelerating the process.

Incoming Call Display
This feature, available on telephones with display fields, provides visual notification to the station user of the calling party's station number or incoming trunk group name. The calling party's name may also be displayed with the station number. The feature is a screeening device to decide whether to answer or divert the call.

Individual Attendant Access
This feature allows users to access a specific attendant console. Each attendant console can be assigned an individual extension number.

Intercom Dial
This feature allows multiple line voice terminal users to gain quickly access to sleect other voice terminal users within an administratered group. Calling voice terminal users lift the handset, press the dial intercom button, and dial the one- or two-digit code assigned to the desired party. The called user received an alerting tone, and the status lamp associated with the Intercom button, if provided, flashes.

Last Number Redialed
This feature stores the last number dialed by the station user and allows the station user to automatically dial the number by using a programmed feature button or feature access code. It simplifies the calling process, reduces misdialed calls, and saves time.

Line Lockout
This feature removes single line voice terminal extension numbers from service when users fail to hang up after receiving dial tone signals, followed by intercept tone signals. The intervals for each tone signal are administrable.

Loudspeaker Paging Access
This feature provides station users or attendants dialing access to voice paging systems. This is useful for paging purposes regardless of the station user's location within the premises environment. It is often used with the call park feature.

Malicious Call Trace
This feature allows a station user to notify a predefined set of station positions that a malicious call is in process. The notified station users can then gather information and data about the call to identify the calling source. The feature is useful when a CLID or ANI is not displayed.

Manual Intercom
This feature allows a station user to call a manual intercom group member by pressing the manual intercom button. All member of a manual itnercom group share a common signaling path. When the manual itnercom button is rpesssed, a special tone burst is sent over the voice terminal speakers of all gropu members. When a group member answers, a speech path is established.

Manual Originating Line Service
This feature connects single line voice terminal users to the attendant automatically when the user lifts the handset. The attendant code is stored in an abbriviated dialing list. When the manual originating line service voice terminal user lifts the handset, the system automatically routes the call to the attendant using the hot line service feature.

Manual Signaling
This feature allows a voice terminal user to signal another voice terminal user. The receiving voice terminal user hears a short burst of tone. The signal is sent each time the button is pressed. If the receiving voice terminal is already being alerted with an incoming call, manual signaling is denied.

Message Waiting
This feature enables multiple line appearance voice terminal users, by pressing a designated button on their own terminals, to light the status lamp associated with the message waiting button at another multiple line apppearance voice terminal. Activating the feature causes the lamp to light on the originating and receiving voice terminals. Either terminal user can cause the lamp to go dark by pressing the button.

Multiparty Conferencing
This feature allows multiappearance voice terminal users to set up multiparty conferences (typically between four to eight station users) without attendant assistance.

Music on Hold
This feature provides music to a party that is on hold, waiting in a queue, parked, or on a trunk call that is being transferred. The music lets the waiting party know that the connection is still in effect. The system provides automatic access to the music source.

[Tips] Inducing delay in VoIP labs

2005-02-18T23:59:00.000+08:00

VOICE OVER IP
Inducing delay in VoIP labs
Tom Lancaster
17 Feb 2005

It's always nice to have the right tools to test your ideas before you implement them, and that goes double for VoIP, but we don't always get what we want. If you have a VoIP lab, odds are that the most unrealistic thing about it is that you're probably getting sub-millisecond response, where your real network has WAN circuits that likely range from 10ms to 120ms or occasionally more. It is of course, much easier to simulate WAN delays with one of those expensive tools, but it can be done the hard way.

If you're in a bind, use a regular router and just configure traffic shaping on it. On a Cisco router, you'll want to use "Generic Traffic Shaping", the details of which are well documented at cisco.com.

Below is an example of the easy, one-line configuration (traffic-shape rate ) and some extended PINGs to show how changing the numbers a bit changes the delay. I configured GTS on the FastEthernet interface of a 2620 and sourced the pings from a 2612 connected to it via a single Catalyst switch.

We start with the basic configuration of the interface and a baseline PING which shows an average of 2ms round-trip response time.

interface FastEthernet0/0
ip address 100.1.1.1 255.255.255.252
duplex auto
speed auto

R4-2610#ping
Protocol [ip]:
Target IP address: 100.1.1.1
Repeat count [5]: 100
Datagram size [100]: 100
Timeout in seconds [2]:
Extended commands [n]:
Sweep range of sizes [n]:
Type escape sequence to abort.
Sending 100, 60-byte ICMP Echos to 100.1.1.1, timeout is 2 seconds:
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Success rate is 100 percent (100/100), round-trip min/avg/max = 1/2/4 ms

Next, I configure a CIR of 32000 bits per second, with a Bc of 1000 bits and Be of 0. The show command explains how the traffic shaping is configured.

interface FastEthernet0/0
ip address 100.1.1.1 255.255.255.252
duplex auto
speed auto
traffic-shape rate 32000 1000 0 1000

R6-2620#show traffic-shape

Interface   Fa0/0
      Access Target    Byte   Sustain   Excess    Interval  Increment Adapt
VC     List   Rate      Limit  bits/int  bits/int  (ms)      (bytes)   Active
-             32000     125    1000      0         31        125       -
R6-2620#

Note that this will allow up to 125 bytes every 31ms, so, let's hop to the other router and send 100 PINGs with a datagram size of 105, plus 20 bytes of IP overhead and see what the times look like:

R4-2610#ping
Protocol [ip]:
Target IP address: 100.1.1.1
Repeat count [5]: 100
Datagram size [100]: 105
Timeout in seconds [2]:
Extended commands [n]:
Sweep range of sizes [n]:
Type escape sequence to abort.
Sending 100, 105-byte ICMP Echos to 100.1.1.1, timeout is 2 seconds:
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Success rate is 100 percent (100/100), round-trip min/avg/max = 1/29/33 ms
R4-2610#

As you can see, we've artificially inserted 27 or so ms of delay into our little network. Back on the 2620, we can see that 94 of our 100 PING packets were delayed. Not bad, eh?

R6-2620#show traffic-shape statistics
                 Acc. Queue Packets   Bytes     Packets   Bytes     Shaping
I/F               List Depth                     Delayed   Delayed   Active
Fa0/0                   0     108       12885     94        11377     no
R6-2620#

One more interesting thing to note is that if we bump the datagram size to 106, which generates PING packets that exceed our GTS Increment of 125 Bytes, it will cause some of our round-trip times to roughly double the average. In other words... voila!: jitter!

R4-2610#ping
Protocol [ip]:
Target IP address: 100.1.1.1
Repeat count [5]: 100
Datagram size [100]: 106   <----- increased to 106
Timeout in seconds [2]:
Extended commands [n]:
Sweep range of sizes [n]:
Type escape sequence to abort.
Sending 100, 106-byte ICMP Echos to 100.1.1.1, timeout is 2 seconds:
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Success rate is 100 percent (100/100), round-trip min/avg/max = 1/29/56 ms
R4-2610#

Back to the delay... I've changed the GTS CIR on the 2620 to half its former rate. And this, predictably, increases the Interval to 62ms, which in turn causes our average PING response time to roughly double.

interface FastEthernet0/0
ip address 100.1.1.1 255.255.255.252
duplex auto
speed auto
traffic-shape rate 16000 1000 0 1000

R6-2620#show traffic-shape

Interface   Fa0/0
      Access Target    Byte   Sustain   Excess    Interval  Increment Adapt
VC     List   Rate      Limit  bits/int  bits/int  (ms)      (bytes)   Active
-             16000     125    1000      0         62        125       -
R6-2620#

R4-2610#ping
Protocol [ip]:
Target IP address: 100.1.1.1
Repeat count [5]: 100
Datagram size [100]: 105
Timeout in seconds [2]:
Extended commands [n]:
Sweep range of sizes [n]:
Type escape sequence to abort.
Sending 100, 105-byte ICMP Echos to 100.1.1.1, timeout is 2 seconds:
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Success rate is 100 percent (100/100), round-trip min/avg/max = 1/58/65 ms
R4-2610#

Cutting the configured CIR to 8000 will in turn yield delays around 120ms. Of course, actual results on your hardware, and any hardware, will vary a bit.

Tom Lancaster, CCIE# 8829 CNX# 1105, is a consultant with 15 years experience in the networking industry, and co-author of several books on networking, most recently, CCSP^TM: Secure PIX and Secure VPN Study Guide published by Sybex.

ENUM in NCNU

2005-02-18T15:27:00.000+08:00

1. 本校的運作機制比較複雜, 若有說明不清楚的地方, 也許當面解釋。
2. 本校校內使用public ip及private ip, private ip是可以在校內流通, 出外才做NAT。所有的電腦連port 80, 都要經過proxy, 未來會使用transparent proxy.
3. 主要負責電信業務的同仁於1月剛離職, 目前由陳順德組長(ext4022 chensd@ncnu.edu.tw)暫代, 另外有一位負責維修的陳家祿(ext4024 gotop@ncnu.edu.tw)
4. enum, 本校是由中興大學分配了25個號碼, 已很多人申請, 本人還不熟,不夠可再提出申請,
5. 雖說enum不能使用NAT, 但是台北已很多地方破了這個問題。

李思宏0963-422090 leeh@ncnu.edu.tw

DNS Entry

2005-02-12T22:47:00.000+08:00

Try to reserve two entries:
1. voip.edu.tw
2. sip.edu.tw