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Module 2: Cisco VoIP Implementations

2.1 Introducing VoIP Networks
2.1.1 Benefits of VoIP Networks
2.1.2 Components of a VoIP Network
2.1.3 Legacy Analog Interfaces in VoIP Networks
2.1.4 Digital Interfaces
2.1.5 Stages for Completing a VoIP Telephone Call
2.1.6 Distributed Call Control
2.1.7 Centralized Call Control
2.2 Digitizing and Packetizing Voice
2.2.1 Basic Voice Encoding: Converting Analog Signals to Digital Signals
2.2.2 Basic Voice Encoding: Converting Digital Signals to Analog Signals
2.2.3 Sampling
2.2.4 Quantization
2.2.5 Digital Voice Encoding
2.2.6 Companding
2.2.7 Common Voice Codec Characteristics
2.2.8 Selecting a Codec Using the Mean Opinion Score
2.2.9 A Closer Look at a DSP
2.3 Encapsulating Voice Packets for Transport
2.3.1 Voice Transport in Circuit-Switched Networks
2.3.2 Voice Transport in IP Networks
2.3.3 Protocols Used in Voice Encapsulation
2.3.4 Voice Encapsulation Codecs
2.3.5 Reducing Header Overhead with cRTP
2.3.6 When to Use RTP Header Compression
2.4 Calculating Bandwidth Requirements for VoIP
2.4.1 Impact of Voice Samples and Packet Size on Bandwidth
2.4.2 Impact of Codecs on Bandwidth
2.4.3 How the Packetization Period Affects VoIP Packet Size and Rate
2.4.4 Data-Link Overhead
2.4.5 Security and Tunneling Overhead
2.4.6 Extra Headers in Security and Tunneling Protocols
2.4.7 Calculating the Total Bandwidth for a VoIP Call
2.4.8 Quick Bandwidth Calculation
2.4.9 Effects of VAD on Bandwidth
2.5 Implementing VoIP in an Enterprise Network
2.5.1 Enterprise Voice Implementations
2.5.2 Deploying CAC
2.5.3 Voice Gateway Functions on a Cisco Router
2.5.4 Cisco Unified CallManager Functions
2.5.5 Enterprise IP Telephony Deployment Models
2.5.6 Cisco IOS Configurations for VoIP
2.6 VoIP Lab Exercises
2.6.1 Lab 2.1 Configure CME using the CLI and Cisco IP Communicator
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