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Cisco 642-432 Exam Syllabus Topics:
Section
Objectives
Integrating Voice over IP into the PSTN
- PSTN Integration
1. Connect a basic VoIP network to the PSTN
2. Explain E.164 addressing
3. Describe digit manipulation
Voice over IP Technologies
- Voice over IP Technologies
1. Explain TDM and statistical multiplexing in telephony
2. Identify VoIP characteristics and transport mechanisms
3. Identify PSTN characteristics and transport mechanisms
4. Compare PSTN and VoIP including call transport, signaling, and bandwidth requirements
5. Differentiate VoIP technologies from PSTN technologies
6. Compare PSTN call signaling with VoIP call signaling
Call Operation and Components in Voice over IP
- Call Operation and Components
1. Explain Quality of Service (QoS)
2. Explain RTP, RTCP, and CRTP
3. Explain voice quality considerations
4. Explain E&M signaling
5. Explain MGCP
6. Explain H.323
7. Describe the basic operation and components of an IP telephony call
8. Describe analog-to-digital conversion
9. Describe companding and compression processes
10. Explain the function and operation of call legs
11. Choose appropriate codecs for different scenarios
12. Explain SIP
13. Explain packetization and codec types
Voice over IP Configuration
- Voice over IP Configuration
1. Configure hunt groups
2. Configure call signaling
3. Configure gateways and gatekeepers
4. Explain digit collection, consumption, and manipulation
5. Configure VoIP and POTS dial peers
6. Configure FXS and FXO ports
7. Configure a router for a basic IP telephony call
8. Explain inbound and outbound dial peer matching
9. Explain default dial peers
10. Configure E&M interfaces
Integrating Voice over IP into Existing PBX Networks
- PBX Integration
1. Select appropriate connection types between PBX and VoIP networks
2. Explain port requirements for PBX connectivity
3. Integrate a basic IP telephony network with an existing PBX
4. Explain signaling requirements for PBX connectivity
Cisco Voice Over IP Sample Questions:
Refer to the exhibit. You have been asked to configure a dial peer on R2 that will match only the extensions of the four telephones attached. Which dial-peer statement will you use?
A. dial-peer voice 1 pots destination pattern 5552[5-6].0
B. dial-peer voice 1 pots destination-pattern 555[2-5][56]
C. dial-peer voice 1 pots destination-pattern 5552.[0-5]0
D. dial-peer voice 1 pots destination-pattern 5552[5-6][05]0
In a VoIP environment when speech samples are framed every 20 ms, a payload of 20 bytes is generated. Assuming a total packet length of 60 bytes, what is the length of the packet header if cRTP is deployed without redundancy checks?
You have set up a complex dial plan using translation rules. The following translation rule has been configured. What output would correspond to the test translation-rule command? translation-rule 1 rule 0 ^0.. 215550210 rule 1 ^1.. 215550211 rule 2 ^2.. 215550212 rule 3 ^3.. 215550213 rule 4 ^4.. 215550214 rule 5 ^5.. 215550215 rule 6 ^6.. 215550216 rule 7 ^7.. 215550217 rule 8 ^8.. 215550218 rule 9 ^9.. 215550210
A. test translation-rule 1 617 The replaced number: 61721555021
B. test translation-rule 1 512 The replaced number: 21555021512
C. test translation-rule 1 910 The replaced number: 21555021910
D. test translation-rule 1 555 The replaced number: 55521555021