ZyXEL Communications Corporation V630 Manuel D’Utilisation

V630 User’s Guide

This chapter discusses the V630’s VoIP > SIP screens.

VoIP (Voice over IP) is the sending of voice signals over the Internet Protocol. This allows you 
to make phone calls and send faxes over the Internet at a fraction of the cost of using the 
traditional circuit-switched telephone network. You can also use servers to run telephone 
service applications like PBX services and voice mail. Internet Telephony Service Provider 
(ITSP) companies provide VoIP service. A company could alternatively set up an IP-PBX and 
provide its own VoIP service.

Circuit-switched telephone networks require 64 kilobits per second (kbps) in each direction to 
handle a telephone call. VoIP can use advanced voice coding techniques with compression to 
reduce the required bandwidth. 

The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol that 
handles the setting up, altering and tearing down of voice and multimedia sessions over the 
Internet.

SIP signaling is separate from the media for which it handles sessions. The media that is 
exchanged during the session can use a different path from that of the signaling. SIP handles 
telephone calls and can interface with traditional circuit-switched telephone networks.

A SIP account uses an identity (sometimes referred to as a SIP address). A complete SIP 
identity is called a SIP URI (Uniform Resource Identifier). A SIP account's URI identifies the 
SIP account in a way similar to the way an e-mail address identifies an e-mail account. The 
format of a SIP identity is SIP-Number@SIP-Service-Domain.

The SIP number is the part of the SIP URI that comes before the “@” symbol. A SIP number 
can use letters like in an e-mail address (johndoe@your-ITSP.com for example) or numbers 
like a telephone number (1122334455@VoIP-provider.com for example).