ZyXEL ADSL Prestige 2602R-61 VoIP 91-006-024004B Manual Do Utilizador

Prestige 2602R Series User’s Guide

Chapter 7 Introduction to VoIP

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This chapter provides background information on VoIP and SIP.

VoIP 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. 

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).