Chapter 9 Voice; Introduction; What You Need To Know - ZyXEL Communications PMG5318-B20A User Manual

9.1 Introduction

This chapter provides background information on VoIP and SIP and explains how to configure your
device's voice settings.
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.

9.1.1 What You Need to Know

The following terms and concepts may help as you read through this chapter.
SIP
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.
SIP Identities
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.
SIP Number
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).
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