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How large should the fragments be to reduce serialization delay to an acceptable level? Well, the real answer lies in an analysis of the delay budgets for your network. From that analysis, you determine the maximum serialization delay you can have on each link.

The delay budget includes many delay components, such as queuing delay, propagation delay, shaping delay, network delay, and serialization delay. Based on that delay budget, you determine how much serialization delay you can afford on a particular link. Figure 8-7 depicts example delay values for various delay components.

Figure 8-7. Review of Delay Components, Including Serialization Delay

Now imagine that you need to configure R1 in the figure to use MLP LFI. You already know that you want a maximum serialization delay of 10 ms, and conveniently, MLP LFI enables you to configure a max-delay parameter. MLP LFI then calculates the fragment size, based on the following formula:

Max-delay * bandwidth

In this formula, bandwidth is the value configured on the bandwidth interface subcommand, and max-delay is the serialization delay configured on the ppp multilink fragment-delay command. For instance, R1 in Figure 8-7 shows a budget for 10 ms of serialization delay. On a 56-kbps link, a 10-ms max-delay would make the fragment size 56,000 * .01, or 560 bits, which is 70 bytes.

Cisco generally suggests a maximum serialization delay per link of 10-15 ms in multiservice networks. Because serialization delay becomes less than 10 ms for 1500-byte packets at link speeds greater than 768 kbps, Cisco recommends that LFI be considered on links with a 768-kbps clock rate and below.

———————————————————————————————————————————————
Note
Earlier Cisco courses, and some other Cisco documents, make the recommendation to set fragment sizes such that the fragements require 10 ms or less. The 10-15 ms recommendation is stated in the current Cisco QoS course.
———————————————————————————————————————————————

The math used to find the fragment size, based on the serialization delay and bandwidth, is pretty easy. For perspective, Table 8-4 summarizes the calculated fragment sizes based on the bandwidth and maximum delay.

Table 8-4. Fragment Sizes Based on Bandwidth and Serialization Delay

—
From Cisco Press – IP Telephony Self-Study Cisco QOS Exam Certification Guide 2nd Edition — LFI

[吃号问题]

左边R1，右边R2
——————————————
R1上：

dial-peer voice 1002 pots
destination-pattern 1002
port 1/0/1

dial-peer voice 1001 pots
destination-pattern 1001
port 1/0/0

dial-peer voice 8888 pots
destination-pattern 9T
port 1/1/0

dial-peer voice 87651000 pots
destianatin-patten 87651000
port 1/0/0

——————————————
R2上:

dial-peer voice 2001 pots
destination-pattern 2001
port 1/0/0

dial-peer voice 87651000 pots
destination-pattern 87651000
no digit-strip
port 1/0/1

——————————————
命令解釋及擴展:
1. 模擬口（pots，包括E1，T1作為PSTN接入時）吃號問題：對於模擬口的dial-peer，要注意的是，在寫destination-pattern這條命令時，該命令下所有的明細號碼將會被吃掉，即不發送出；

2. “T”表示所有数字,包括沒有，“.”表示一個任意，如果用9T，那麼9會被吃掉，T（任意）會被發出去,如果是S口直接連接終端電話機,被吃掉也無所謂；上述配置中R2上由於S口連接到了R1的O口，在這裡寫了destination-pattern 87651000，那麼87651000會被吃到，R1將不會收到被叫號碼的資訊，我們可以通過在dial-peer下的兩種命令來解決：
1）no digit-strip：精確匹配的號碼不會被吃掉
2）Prefix 87651000：表示補回87651000後發出去

3. 9T和9.T有什麼區別?
9T：當你撥完 9後，什麼都不撥，timeout後信令會傳出去（off-hook給R2， R2也會回應你，，鏈路被佔用，R2會提供2次撥號聲）
9.T：當你撥完 9後，什麼都不撥，timeout後線路不會被觸發，線路不會被佔用。（命中兩位才會命中路由，才會發出信令）

4. 兩次回鈴音的解釋：（有ring必有ringback）
R2後的2001打87651000後，R2的S口認為連接的是一台電話，發出ring，相應的向1/0/0發送ringback（第一聲回鈴音）；R1收到後同樣有一個ringback（第二聲回鈴音）

[CAS]

controller T1 3/0
framing sf // 成幀位，T1為“sf（12幀一個序列）”和“esf” （24幀一個序列）
linecode ami // 具體的數字型號（010101.。）變成電平的方式，
clock sou line
ds0-group 0 timeslots 1-3 // (當配置資料時，利用 channel-group 1 timeslots 1-3)

!
dial-peer voice 9 pots
destination-pattern 9T
port 3/0:0
!

在語音中，T1的CAS信令叫做RBS（Robbed-Bit Signaling）即“奪位”信令，因為沒有專門的信令通通道，所以就需要在語音通道裡借位來傳輸信令，由於T1的同步機制有兩種：超幀（SF，每12個frame同步一次）和擴展超幀（ESF，每24個frame同步一次）.

[CSS]

Debug isdn q931
1) 配置ISDN以及外線
isdn switch-type primary-ni (T1一般為ni,,E1為net-5)
Client(config)#isdn switch-type ?
primary-4ess Lucent 4ESS switch type for the U.S.
primary-5ess Lucent 5ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss DPNSS switch type for Europe
primary-net5 NET5 switch type for UK, Europe, Asia and Australia
primary-ni National ISDN Switch type for the U.S.
primary-ntt NTT switch type for Japan
primary-qsig QSIG switch type
primary-ts014 TS014 switch type for Australia (obsolete)
!

controller t1 2/0
framing sf
linecode ami
pri-group timeslots 1-12 ,24 (第24個chan為信令位，你不配會自動配上去的，注意和下面23对比）

注意：在ISR上，如在28和38路由器上需要添加下列命令
network-clock-participate slot 2
network-clock-select 1 T1 2/0
！

dial-peer voice 9 pots
destination-pattern 9T
port 2/0:23 // 這個23就是信令channel（路由器在這裡按0~23算的，注意和上面24的區分）
!

R2配置（R2做為CO需要一些額外的配置，如下）

dial-peer voice 4321 pots
destination-pattern 43212001
port 1/0/0
!
isdn switch-type primary-ni
！
controller t1 3/0
framing sf
linecode ami
pri-group timeslots 1-12
!
interface serial 3/0:23
isdn protocol-emulate network // 發出無編號的確認資訊， 服務商端的配置
isdn incoming-voice voice // 遇到voice交給voice處理（默認會加上的）
！
dial-peer voice 8765 pots
destination-pattern 87651…
forward-digits all // 同 no digit-strip (如果寫 port 3/0:23 forward-digits 4 就表示保留最後4位)

2) 二次撥號問題解決
從R1撥向R2時，先信令，然後將被叫號碼發給R2，這時發現1001上收到二次撥號聲但撥號沒有成功，當再一次撥好後成功 R1的Debug q931：

分析：CCS：23B+1D；對於PRI的原始行為是監聽B通道上的DTMF，但PRI在連接階段是沒有DTMF的，只有信令，所以對方路由器最初沒有在B通道上監聽到相應的號碼所以返回重播的二次撥號聲，事實上這時的號碼在D通道通過DNIS在傳遞（看上面的Debug）。
解決方法：我們在R2上配置如下命令，使R2在D通道上監聽號碼（DID模組：分機和直線關聯，直線撥入）DID模組：兩個功能一是可以定義在D通道上監聽DNIS號碼，另外一個是直線撥入 建議在配置ISDN時配上DID：

R2：
dial-peer voice 1 pots // 隨意定義一條
incoming called-number . // 配置呼入時的被叫號碼，注意有個點
direct-inward-dial
上述的命令表示監聽D通道的DNIS，匹配任何就路由。
R1:
dial-peer voice 1 pots // 隨意定義一條
incoming called-number .
direct-inward-dial
注意有個點，表示有1位就命中了，如果不是”.“是888那麼這個dial-peer就無效，繼續檢查下一個指標ANI（主叫號碼）
注意：T表示任意，包括沒有。

Topo:

SIP — CCM —— CME — SCCP
CCM:100.100.19.181
CME:100.100.19.190
CCM’s DN:1XXX
SCCP’DN on CME:2004 and 2002

0. Setup a Trunk to CME Router (Inter-Cluster Trunk (Non-Gatekeeper Controlled) or SIP Trunk)
(add the CME as H323 GW will not work here, why?)

1. Add a Route Pattern 2XXX, point to CME Router

2. Unzip files its-CISCO.2.0.3.0.zip and import all of them to CME Router’s flash

3. Configure service aa on router:
—————————————-
!
application
service aa flash:its-CISCO.2.0.2.0.tcl
param operator 2002
paramspace english index 0
paramspace english language en
paramspace english location flash:
param interDigitTimeout 3
paramspace english prefix en
param max-extension-length 4
param aa-pilot 2004
param welcome-prompt _welcome.au
!
!
—————————————-

4. Register one SCCP to this Router under CME mode, with two DN: 2004 and 2002, 2004 as aa-pilot, 2002 as operator.

5. Configure dial-peer for service aa:
!
dial-peer voice 2004 voip
service aa
incoming called-number 2004
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
no vad
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 3
!

6. Configure dial-peer for calls to CCM:
!
dial-peer voice 1 voip
destination-pattern 1…$
session protocol sipv2 // using sip GW
session target ipv4:100.100.19.181
voice-class codec 1
no vad
!

Now, the sip phone under CCM can dial 2004 and hear the welcome_prompt, and can press 0# for operator, and dial any extension in CCM.

Cisco Catalyst 3750 QoS Configuration Examples
Introduction
Prerequisites
Requirements
Components Used
Conventions
QoS Overview
Cisco Catalyst 3750 Switch without QoS
Cisco Catalyst 3750 Switch QoS Features
Ingress QoS Features
Default Ingress QoS Configuration
Classification and Marking
Policing
Congestion Management and Avoidance
Egress QoS Features
Egress QoS Commands
Default Configuration
Queuing, Dropping and Scheduling
Cisco Support Community – Featured Conversations
Related Information

Understanding Delay in Packet Voice Networks
Introduction
Basic Voice Flow
How Voice Compression Works
Standards for Delay Limits
Sources of Delay
Coder (Processing) Delay
Packetization Delay
Serialization Delay
Queuing/Buffering Delay
Network Switching Delay
De-Jitter Delay
Build the Delay Budget
Single-Hop Connection
Two Hops on a Public Network with a C7200 that Acts as a Tandem Switch
Two-Hop Connection over a Public Network with a PBX Tandem Switch
Two-Hop Connection over a Private Network with a PBX Tandem Switch
Effects of Multiple Compression Cycles
Considerations for High-Delay Connections
Cisco Support Community – Featured Conversations
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Understanding IP Telephony Protocols