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Voice CCIE直通车
适合群体：从零基础开始,一直到通过Vocie CCIE.希望将来从语音网络服务工程师的工作.
学习过程: CCNA+CCNP+CCVP+CCIE,培训费已经包括书,实验,理论.但不含思科CCIE笔试和实验考试费(思科笔试费在3000左右，实验考试费在10800左右人民币,需自己另交思科) 如会员自己放弃不学CCNA和CCNP课程总价格不变.
学习周期:6个月左右(其中CCNA 4天+ CCNP 1轮的学习共1个半月+CCVP+Vocie CCIE理论和实验+考前冲刺)
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| VOICE CCIE 课程内容 |
| 一、园区网IP语音的VLAN设计： |
Basic Campus Design
DHCP, TFTP
Catalyst voice and data VLAN configuration
Catalyst VTP configuration | | |
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| 二、呼叫管理器和简化版呼叫管理器的系统设置，IP电话注册和配置： |
Call Manager and Call Manager Express Configuration
Phone registration
Phone configuration | | |
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| 三、IP语音网关和关守，传统语音信令、IP语音信令： |
Voice Gateway and Signaling
Analog and digital voice protocols: analog FXS, FXO-M1, T1, E1, PRI, CAS, R2
VoIP protocols: H323, MGCP, SCCP, SIP, RAS | | |
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| 四、IP语音呼叫路由，路由策略，呼叫号码转换： |
Call Routing
CCM route patterns (@ wildcard not tested)
CCM route preference and redundancy
IOS dial peers
Digit manipulation and translation | | |
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| 五、IP语音编码： |
Voice CODEC
G711ulaw, G711alaw, G729, G723 | | |
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| 六、呼叫准入控制，基于呼叫管理器和基于关守的带宽调控： |
Call Admission Control
Location-based
RAS-based | | |
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| 七、高可用性策略、远端电话存活功能和自动重路由功能： |
High Availability Features
SRST
AAR | | |
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| 八、媒体资源管理、会议桥、编码转换器、媒体终结点、保持音乐设置： |
Media Resource Management
Conference bridge software and hardware
Transcoder
MTP
MOH | | |
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| 九、QOS设计、基于二层和三层的QOS： |
QoS Considerations
L2/L3 classifications and policing
Queuing mechanisms
LFI
Catalyst switch QoS | | |
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| 十、统一消息平台设计、语音邮箱的使用、呼叫流程的设置： |
Unified Messaging
Unity voicemail integration
Unity administration | | |
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| 十一、简化版联络中心/呼叫中心的平台设计、脚本应用： |
CRS/IPCC Express Application
Default script configuration and integration
Custom script configuration and integration | | |
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| 十二、呼叫管理器的增值应用、移动分机、话务总控、话务助理： |
Call Manager Voice Applications
Any native applications to Call Manager, examples are extension mobility, attendant console, IPMA | | |
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| 十四、企业黄页服务： |
| Directory Services and Integration | | |
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| 十五、传真： |
Fax
Fax pass-through
Fax relay | | |
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| 十六、前期基础必修课程 - CCVP |
| VOICE CCIE 的前期基础必修课程为VOICE CCVP，包括以下： |
| A. |
Cisco Voice Over IP (CVOICE) – CISCO IP语音
Introducing Voice Over IP
Voip Network Technologies
VoIP Network Architectures
Building Scalable Dial Plans
Calculating Bandwidth Requirements
Allocating Bandwidth for Voice and Data Traffic
Considering Security in VoIP Networks
Configuring Voice Networks
Configuring Voice Ports
Adjusting Voice Interface Settings
Configuring Dial Peers
Configuring Voice Port Connections
VoIP Signaling and Call Control
Introducing Signaling and Call Control
Introducing H.323
Deploying and Configuring H.323
Configuring SIP
Configuring MGCP
Comparing Call Control Models
Improving and Maintaining Voice Quality
Designing for Optimal Voice Quality
Implementing CAC
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| B. |
Cisco Voice Over IP Fundamentals (CVF)–CISCO传统语音基础
Traditional Voice Telephony
Requirements of Voice on an IP Network
VoIP Benefits and Services |
| C. |
Implementing Cisco Unified CallManager Part 1 (CIPT1)–CISCO呼叫管理器第一部分
Module 1 - Getting started with Cisco CallManager Release 5.0
Lab 1-1 Installing Cisco CallManager 5.0
Module 2 - Administering Cisco CallManager Release 5.0
Lab 2-1 Performing Platform Administration
Lab 2-2 Performing General Administration
Lab 2-3 Configuring CallManager 5.0 Basic Settings
Lab 2-4 Configuring Cisco CallManager to Support IP Phones
Module 3 - Deploying Cisco CallManager Release 5.0 Endpoints
Lab 3-1 Configuring Auxiliary or Voice VLANs
Lab 3-2 Configuring SIP Endpoints
Lab 3-3 Using the Bulk Administration Tool
Module 4 - Deploying a Dial Plan on CallManager Release 5.0
Lab 4-1 Configuring Voice Gateways
Lab 4-2 Configuring Cisco CallManager Trunks
Lab 4-3 Configuring Basic Dial Plan Elements
Lab 4-4 Configuring Hunt Groups and Call Coverage
Lab 4-5 Configuring Complex Dial Plan Elements
Module 5 - Implementing Multi-site Deployments
Lab 5-1 Configuring Call Admission Control in Cisco CallManager 5.0
Lab 5-2 Configuring SRST
Module 6 - Enabling Features and Services
Lab 6-1 Configuring Media Resources
Lab 6-2 Configuring User Features Part 1
Lab 6-3 Configuring Extension Mobility
Lab 6-4 Configuring Additional Features
Lab 6-5 Configuring Cisco CallManager Attendant Console
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| D. |
Implementing Cisco Unified CallManager Part 2 (CIPT2) – CISCO呼叫管理器第二部分
Course Introduction
Securing IP telephony
Enabling IP video telephony
Monitoring and Managing IP telephony
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| E. |
Implementing Cisco Voice Gateways and Gatekeepers (GWGK) - CISCO网关和关守
* Gateway Deployments Introducing Gateways Implementing H.323 Gateways Implementing MGCP Gateways Implementing SIP Gateways Implementing SRST Gateways
* Cisco IP Communiczations PSTN and PBX Imtegrations Introducing PSTN and PBX Trunks Implementing Analog Trunks Implementing CAS Trunks Implementing BRI and PRI Trunks Implementing QSIG Trunks
* Dial Plans on IOS Gateways Dial Plan Overview Implementing PSTN Dial Plans on IOS Gateways Implementing PSTN Dial Plans on IOS Gateways Implementing Multi Site Dial Plans on IOS Gateways Implementing Calling Privileges on IOS Gateways Implementing RSVP Based CAC
* Advanced Gateway Features Implementing Media Resources using IOS Gateway DSPs Implementing FAX Support on IOS Gateways Call Applicaitons on IOS Gagteways
* Gatekeeper Deployments Introducing Deployments Implementing Basic Gatekeeper Functionality Implementing Gatekeeper Call Adminsion Control Implementing Advanced Gatekeeper Functionality Implementing Gatekeeper Redundancy
* IP-IP Gateway IP-IP Gateway Overview Implementing IP-IP Gateways
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| F. |
IP Telephony Troubleshooting (IPTT/TUC) – CISCO IP电话故障排除
Troubleshooting Methodology
Analyzing and Isolating Problems
Common Unified Communications Problems Symptoms
Fixing Problems
Services, Resources and Tools for Troubleshooting
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| G. |
Implementing Cisco Quality of Service (QOS) – CISCO QOS - 此项为自修课 | | |
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