Voice over IP with SIP
This course focuses on theoretical and practical principles of Real-Time Media over Internet Protocol coupled with SIP (Session Initiation Protocol), H.323, RTP and RTCP. Delegates will configure basic parameters on a SIP Server and test calls across the network.
By the end of the course you will be able to:
- Understand how VoIP and it’s associated protocols fit in with the existing networking protocol models.
- Be able to explain the reasons for the use of VoIP.
- Describe the potential benefits of VoIP
- Understand voice quality issues associated with VoIP
- State the additional protocols that make VoIP possible
- Understand the differences between H.323 and SIP
- Understand Quality of Service (QoS) and what it means
- Appreciate how IP Multicasting plays a part in Real-time media delivery
- Understand the differences between Circuit Switches and Packet Switched Voice.
- Configure a simple VoIP application on a windows PC and make Phone Calls
- Set up a SIP Server for registration of Client Devices
- Configure a SIP Converter or SIP Phone
- Configure a SIP Server / IP PBX at multiple sites and make calls across a WAN.
- Configure and test basic QoS parameters and Test Voice Quality
A basic understanding of Data Networking and Internet Protocol (Version 4).
Network and Telecommunications technicians and engineers. Also network support staff and project managers working with networks that have or are about to have a VoIP element.
Introduction to VoIP
Voice Encoding Schemes
Protocols – Network and Transport Layers
Protocols – Data Link and Physical Layers
VoIP Support Services
Real Time Protocols
SIP – Session Initiation Protocol
Cisco VoIP Example (Brief)
QoS – Quality of Service
VoIP Phone / Adapter and IP PBX Configuration