ZyXEL Communications P-2602HWLNI, User Manual, Page: 169 / 496

P-2602HWLNI User’s Guide
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H A P T E R
11
Voice
This chapter provides background information on VoIP and SIP and explains how to configure
your device’s voice settings.
11.1 Introduction to VoIP
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.
11.2 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.
11.2.1 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.
11.2.1.1 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).