User manual

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Chapter 8 Voice
This chapter first describes the various options for configuration of the SIP voice
service. It then provides detailed instructions for making telephone calls using VoIP
(Voice over IP) or PSTN (Public Switched Telephone Network) services.
Session Initiation Protocol (SIP) is a peer-to-peer protocol used for Internet
conferencing, telephony, events notification, presence and instant messaging.
SIP 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.
NOTE: The SIP standard is set by the Internet Engineering Task Force (IETF).
The SIP standard defines the following agents/servers:
 User Agents (UA) - SIP phone clients (hardware or software)
 Proxy Server – relays data between UA and external servers
 Registrar Server - a server that accepts register requests from UA
 Redirect Server – provides an address lookup service to UA
The following subsections present Basic, Advanced and Debug SIP screens.
Each screen provides various options for customizing the SIP configuration.