ZyXEL Communications Corporation P8802T Manuel D’Utilisation

 Chapter 15 Voice

Basic Home Station VDSL2 P8802T User’s Guide

This section contains background material relevant to the VoIP screens.

VoIP is the sending of voice signals over 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.

phone port

This is the phone port on which you received the call.

Missed means the call was unanswered.

phone number

This is the SIP number that called you.

duration

This displays how long the call lasted.

Table 79   VOIP > Call History > Incoming