VAB: Voice over IP
This course describes the components, protocols and operation of telephony systems that operate over Internet Protocol (IP) networks with emphasis on the Session Initiation Protocol (SIP).
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Target audience:
Those responsible for the design and maintenance of private or public telephony systems.
Course prerequisites:
An understanding of the basic principles of telephony. It is recommended that the PTT online course "Introduction to telephony" is studied before attempting this "Voice over IP" course.
Prior knowledge of Ethernet and IP networks will also be beneficial. The PTT online courses "Ethernet networks" and "Internet protocols" cover these topics.
Course objectives:
By the end of this course you will be able to:
- describe the role of the various stages of processing of speech in a VoIP system.
- explain the factors that determine the bandwidth requirement of a VoIP system.
- describe how the real-time transport protocol (RTP) minimises the effects of network impairments.
- describe the role of a user agent, proxy server, register, redirect server, location server, and PSTN gateway in a VoIP system using the session initiation protocol (SIP).
- describe the role and format of, and facilities provided by, the various signalling messages employed in systems using SIP.
- explain the barriers to VoIP provision with reference to network address translation (NAT) and firewalls.
- describe the role of, and facilities provided by, a session border controller (SBC).
- explain the various methods of overcoming the barriers to VoIP provision including STUN, port forwarding, and the use of an SBC.
- explain the use of digital certificates, transport layer security (TLS), and secure RTP (SRTP) in ensuring secure voice communications.
- explain how SIP-based fixed and mobile public telephony services provide an assured quality of service.
- describe the provision and benefits of a hosted ("cloud-based") VoIP service with reference to the role of, and methods of providing, a SIP trunk.
- explain how cloud-based telephony and video conferencing services can be integrated with existing private VoIP systems and provide alternative routes for external calls.
Number of modules: 5
Approximate study time: 6 hours
Download and view the course specification in Adobe PDF format
Delivery platform requirements:
To access the course, a computer running a browser such as Google Chrome, Safari etc is required. The computer should have Internet access. A screen resolution of at least 1024x768 is necessary.
