As PABX vendors implement the latest and greatest VoIP strategies in their PABX systems, an unfortunate design problem often occurs because the phone system is implemented on top of an existing data network system designed for PC and server use, but not for IP phone use.
The Designing & Implementing a Voice-Enabled IP Network course has been designed with three primary goals:
Anyone planning to upgrade or considering expansion of their PABX system in the near future should take advantage of this course.
Students will design and implement a secure network to simulate the introduction of an NEC IP PABX. To accomplish this, they will configure Cisco routers and switches as well as NEC IP phones, ensuring IP Voice traffic is securely separated from PC IP traffic using VLAN technology and routing.
During the course students will also configure IP phones and observe the various networking standards at work, and use a PC based PABX (Asterisk) as well as the NEC PABX to place VoIP calls. They will observe the effects of delay and jitter caused by combining voice and data on an IP network, then configure Cisco routers and switches to minimise these problems. VoIP network design strategies will be discussed including different vendor offerings and implementations.
Students will receive a copy of Theodore Wallingfords popular book Switching to VoIP around which part of the course is based.
Who Should Attend:
The Designing & Implementing a Voice-Enabled IP Network is ideal for anyone who is planning to replace or upgrade their PABX in the future.
Prerequisites:
Format:
40% lecture, 60% exercises and hands-on labs with two softphones, two NEC DT700 IP phones, an Asterisk IP PABX, one Cisco switch and one Cisco router per group of two students.
Course Content:
Voice and Data: Two Separate Worlds?
The PSTN
Key Systems and PBXs
Limits of Traditional Telephony
VoIP in the Home
VoIP in Business
VoIPs Changing Reputation
Voice over Data: Many Conversations, One Network
The Layers of a VoIP Network
Distributed Versus Centralised
Linux as a PBX
Free Telephony Software
Installing Legacy Interface Cards
Monitoring Asterisk
Circuit-Switched Telephony
Components of the PSTN
Customer Premises Equipment
Time Division Multiplexing
Point-to-Point Trunking
Legacy Endpoints
Dial-Plan and PBX Design
Enterprise Telephony Applications
Application Terminology
Basic Call Handling
Administrative Applications
Messaging Applications
Advanced Call-Handling Applications
CTI Applications
Replacing the Voice Circuit with VoIP
The Dumb Transport
Voice Channels
Project: Set Up Custom Codec Selection and Enable an Independent Call Path
Replacing Call Signalling with VoIP
VoIP Signalling Protocols
H.323
SIP
IAX
MGCP
Cisco SCCP
Heterogeneous Signalling
VoIP Readiness
Assessing VoIP Readiness
Business Environment
Network Environment
Implementation Plan
Quality of Service
Latency, Packet Loss, and Jitter
CoS
802.1q VLAN
Quality of Service
Residential QoS
Voice QoS on Windows
Best Practices for Quality of Service
Troubleshooting Tools
VoIP Troubleshooting Tools
The Three Things Youll Troubleshoot
SIP Packet Inspection
Interoperability
Project: Trace Both Ends of a Call Setup with Log Comparison
When, Not if, You Have Problems...
Simulating Media Loads
PSTN Trunks
Dial-Tone Trunks
Routing PSTN Calls at Connect Points
Timing Trunk Transitions
Network Infrastructure for VoIP
Legacy Trunks
VoIP Trunks
Project: Build an Interactive Directory on a SIP Display Phone
Traditional Apps on the Converged Network
Fax and Modems
Fire and Burglary Systems
Surveillance Systems and Videoconferencing
Voice Mail and IVR
Emergency Dispatch
What Can Go Wrong?
Common Problem Situations
VoIP Vendors and Services
Softphones and Instant Messaging Software
Skype
Other Desktop Telephony Software
Developer Tools and SoftPBX Systems
VoIP Service Providers
Telephony Hardware Vendors