Zyxel IES-1000 EE [103/368] Introduction to sip

Zyxel IES-1000 EE [103/368] Introduction to sip
VOP1224-61 User’s Guide
103
CHAPTER 17
VoIP
This chapter shows you how to configure the Voice over Internet Protocol (VoIP)
features on your VOP.
17.1 SIP Overview
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.
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 VOP connects POTS (Plain Old Telephone System) end-user telephone
subscribers to the IP network by converting the analog voice signal into data
packets and transmitting them over the network.
17.1.1 Introduction to 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.
17.1.1.1 SIP Registration
Each VOP is an individual SIP User Agent (UA). To provide voice service, it has an
IP address for SIP and RTP protocols to communicate with other servers.

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