This 2 day Instructor led course is designed for design and support specialists with a working knowledge of voice and data who need the fundamentals of Voice over IP technologies.
The knowledge and skills that a learner must have before attending this course are as follows:
- Voice Telecommunications
- Internet Protocol (IP) Fundamentals
Major Components of the VoIP Network
- VoIP Network
- Local Area Network Infrastructure
- Wide Area Network Infrastructure
- Security Issues
- Analog to Digital Conversion
- Pulse Code Modulation
- Packet Switching
- Voice Packet
Transport and Session Layer Technologies
- Transmission Models
- Transport and Session Layer Internet Protocols
- CODEC Selection
- Common Models for Voice Quality
- Bandwidth Overview
- Calculating bandwidth without Layer 2
- Layer 2 Data Link Overhead
- Calculating bandwidth with Layer 2
- Calculating Effective bandwidth with Voice Activity Detection
Quality of Service
- Methods to Achieve Quality of Service
- Ethernet 802 Standards
- Queuing Mechanisms
- A layered approach
Traffic Convergence Issues
- Low Speed WAN Connection Issues
- Common Ethernet Network Issues for LAN Environment
- Describing WLAN Security Standards
VoIP Standardization and Signaling Protocols
- H.323 Overview
- Session Initiation Protocol (SIP)
- Comparing H.323 and SIP
- Interworking H.323 and SIP
- Other Signaling Protocols
- Readiness Audit
- Network Diagram
- Link Types and Speeds
- Power and Wiring
- Quality of Service
- Packet Loss and delay
- Protocol Considerations
- Define the key infrastructure considerations to support the addition of VoIP traffic in a Local Area Network, Wide Area Network, and Wireless environments.
- Apply knowledge of how voice is sampled and converted into IP packets to determine the appropriate CODEC and packetization interval required to meet customer VoIP bandwidth requirements
- Compare and contrast transport models, such as Voice over IP, Voice over Frame Relay, and Voice over AT.
- Describe standard voice quality measurement models, such as the Mean Opinion Score and the E-Model.
- Apply knowledge of the attributes of Real-Time Protocol to identify why it is ideal for handling packetized voice in an IP telephony environment.
- Compare the unique attributes of User Datagram Protocol and Transmission Control Protocol.
- Choose the appropriate CODEC given customer bandwidth and voice quality performance requirements.
- Determine bandwidth requirements based upon call volume and CODEC selection.
- Choose the appropriate method to implement Quality of Service given customer voice quality requirements.
This course is designed for:
- Avaya VOIP Technicians and Engineers