The
following topics are general guidelines
for the content likely to be included
on the CIPT Specialist CVOICE Exam 9E0-431. However,
other related topics may also appear on
any specific delivery of the exam (excerpts
from Cisco Certification web site).
1. Introduction to Packet Voice Technologies
1.1. Explain traditional telephony networks
1.2. Describe packetized voice networks
1.3. Describe IP data networks
2. Analog and Digital Voice Connections
2.1. Recognize the basics of analog and
digital voice
2.2. Explain the processes and standards
of voice digitalization, compression,
and digital signaling
2.3. Describe signaling methods including
ISDN voice interfaces, signaling between
PBXs, Common Channel Signaling Systems,
and interworked signaling systems
2.4. Describe the standards, applications,
and issues of FAX and modem usage over
a VoIP network
3. Configuring Voice Interfaces
3.1. Configure analog and digital voice
interfaces
3.2. Configure analog and digital voice
ports for optimal voice quality
3.3. Given a scenario, identify how to
customize and verify analog port operations
3.4. Given a scenario, identify how to
create, customize, and verify digital
port operations
4. Voice Dial Peers
4.1. Describe call flows as they relate
to inbound and outbound dial peers
4.2. Describe the need for and application
of voice dial peers
4.3. Describe the use and application
for various special purpose connections
on Cisco telephony equipment
4.4. Given a scenario, determine appropriate
method of digit forwarding and manipulation
4.5. Given a scenario, identify how to
create a trunk connection for calls between
two PBXs
4.6. Given a scenario, identify the appropriate
information from show and debug commands
to diagnose a fault with digit forwarding
and manipulation, with PLAR/PLAR OPX
and trunk connections between two PBXs
5. Introduction to Voice over IP
5.1. Describe the fundamentals of VoIP
and identify challenges and solutions
regarding its implementation
5.2. Discuss the differences and similarities
between VoIP and Voice over other technologies,
such as Frame Relay or ATM
5.3. Describe the roles of Gateways in
integrating VoIP with the traditional
voice technologies found in enterprise
and service provider networks
5.4. Define the VoIP protocol stack,
its applied headers, and the use of RealTime
Transport Protocol compressed (cRTP)
to reduce header size
5.5. List the bandwidth requirements
for various codecs and data links and
methods to reduce bandwidth consumption
5.6. Given a scenario, describe and configure
proper use of dial peer CODEC parameters
5.7. Given a scenario, verify basic call
setup through debug commands
5.8. Given a scenario, identify appropriate
show and debug commands to monitor and
troubleshoot the connections
6. Voice over IP Signaling and Call
Control
6.1. Describe the need for and the operation
of various signaling and call control
models
6.2. Describe call control services such
as H.323, Session Initiation Protocol
(SIP), and Media Gateway Control Protocol
(MGCP)
6.3. Explain the functional components
of SIP and the interactions between them
6.4. Explain the functional components
of MGCP and their interactions
6.5. Distinguish between centralized
and decentralized call control
6.6. Define the functional components
of H.323 and explain the interactions
between them
6.7. Compare the H.323, SIP, and MGCP
call control models
6.8. Given a scenario, identify how to
configure single zone and multizone H.323
Gatekeeper environments for VoIP scalability
6.9. Given a scenario, identify how to
use debug and show commands to monitor
the status and progress of call setup
procedures in an H.323 environment
6.10. Given a scenario, identify how
to configure dial peers to use SIP call
control procedures to setup VoIP calls
6.11. Given a scenario, identify how
to configure your routers as MGCP Residential
Gateways and have the routers
6.12. Given a scenario, identify how
to use debug commands to analyze the
interactions between MGCP gateways and
a Call Agent
7. Use show commands to view the status
of MGCP endpoints, connections, and calls
8. Improving and Maintaining Voice Quality
8.1. State the purpose for Voice Quality
Measurement including codec choice,
which affects quality
8.2. List the challenges of transporting
realtime voice in a non-realtime IP internetwork
8.3. Describe Quality of Service (QoS)
functional areas and tools and the effect
of network design on QoS
8.4. Describe jitter, its overall affect
on voice quality, and how to overcome
it
8.5. Describe delay, its overall affect
on voice quality, how it is measured,
and how to overcome it
8.6. Describe the QoS tools used in a
campus network
8.7. Describe the QoS tools used in a
WAN
8.8. Describe Call Admission Control
tools and the function and operation
of Resource Reservation Protocol (RSVP)
8.9. Calculate the busy hour bandwidth
allocation for both voice and data traffic
of an existing network
8.10. Given a scenario, identify how
to implement quality improvements on
low speed links with QoS features such
as fragmentation, interleave, and Frame
Relay traffic shaping
8.11. Given a scenario, identify how
to improve voice quality end-to-end
8.12. Given a scenario, identify how
to confirm by way of testing that each
QoS feature is contributing to overall
improvements in voice quality
8.13. Given a scenario, identify how
to implement call admission control by
setting the dial peer maximum connections
8.14. Given a scenario, identify how
to confirm by way of testing that call
admission control techniques can effectively
limit the number of VoIP calls
9. Scalable Numbering and Applications
9.1. Implement a scalable numbering plan
in a VoIP network
9.2. Identify new and evolving applications
and cost-saving ideas that capitalize
on the convergence of voice and data
in an IP internetwork
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