User`s manual
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Chapter 7 Voice
This chapter first describes the SIP and Dial Plan configuration screens. The
last section (7.3 Telephone Calls) describes how to use the VoIP (Voice over IP)
and PSTN (Public Switched Telephone Network) voice services.
7.1 SIP
Session Initiation Protocol (SIP) is a signaling protocol for Internet conferencing,
telephony, presence, events notification and instant messaging. It is the Internet
Engineering Task Force's (IETF's) standard for multimedia conferencing over IP.
It is designed to address the functions of signaling and session management
within a packet telephony network. Signaling allows call information to be carried
across network boundaries. Session management provides the ability to control
the attributes of an end-to-end call.
Session Initiation Protocol is a peer-to-peer protocol.
There are four components in the SIP standard:
(a) User Agents (UA) - SIP phone clients (hardware or software)
(b) Proxy Server – relays data between UA and external servers
(c) Registrar Server - a server that accepts register requests from UA
(d) Redirect Server – provides an address lookup service to UA