P-2302R Series VoIP Station Gateway User's Guide
Table Of Contents
- P-2302R Series
- User’s Guide
- Copyright
- Federal Communications Commission (FCC) Interference Statement
- Safety Warnings
- ZyXEL Limited Warranty
- Customer Support
- Table of Contents
- List of Figures
- List of Tables
- Preface
- Introducing the Prestige
- Introducing the Web Configurator
- Wizard Setup
- System Screens
- LAN Setup
- WAN Screens
- Introduction to VoIP
- VoIP Screens
- Phone
- Phone Book
- Phone Usage
- Network Address Translation (NAT) Screens
- Static Route
- Firewall
- Content Filtering
- Remote Management Screens
- Universal Plug-and-Play (UPnP)
- Logs
- Bandwidth Management
- 19.1 Bandwidth Management Overview
- 19.2 Bandwidth Classes and Filters
- 19.3 Proportional Bandwidth Allocation
- 19.4 Application-based Bandwidth Management
- 19.5 Subnet-based Bandwidth Management
- 19.6 Application and Subnet-based Bandwidth Management
- 19.7 Scheduler
- 19.8 Maximize Bandwidth Usage
- 19.9 Bandwidth Borrowing
- 19.10 Configuring Summary
- 19.11 Configuring Class Setup
- 19.12 Configuring Monitor
- Maintenance
- Introducing the SMT
- General Setup
- WAN Setup
- LAN Setup
- Internet Access
- Remote Node Configuration
- Static Route Setup
- Network Address Translation (NAT)
- Enabling the Firewall
- Filter Configuration
- SNMP Configuration
- System Information and Diagnosis
- Firmware and Configuration File Maintenance
- 33.1 Filename Conventions
- 33.2 Backup Configuration
- 33.2.1 Backup Configuration
- 33.2.2 Using the FTP Command from the Command Line
- 33.2.3 Example of FTP Commands from the Command Line
- 33.2.4 GUI-based FTP Clients
- 33.2.5 TFTP and FTP over WAN Management Limitations
- 33.2.6 Backup Configuration Using TFTP
- 33.2.7 TFTP Command Example
- 33.2.8 GUI-based TFTP Clients
- 33.3 Restore Configuration
- 33.4 Uploading Firmware and Configuration Files
- System Maintenance
- Remote Management
- Call Scheduling
- Troubleshooting
- 37.1 Problems Starting Up the Prestige
- 37.2 Problems with the LAN Interface
- 37.3 Problems with the WAN Interface
- 37.4 Problems with Internet Access
- 37.5 Problems with the Password
- 37.6 Problems with the Web Configurator
- 37.7 Problems with a Telephone or the Telephone Port
- 37.8 Problems with Voice Service
- 37.9 Pop-up Windows, JavaScripts and Java Permissions
- Product Specifications
- Wall-mounting Instructions
- Setting up Your Computer’s IP Address
- IP Subnetting
- PPPoE
- Triangle Route
- SIP Passthrough
- Index

P-2302R Series User’s Guide
Chapter 7 Introduction to VoIP 90
CHAPTER 7
Introduction to VoIP
This chapter provides background information on VoIP and SIP.
7.1 VoIP Introduction
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. A company could alternatively set up an IP-PBX and
provide it’s own 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.
7.2 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.
7.2.1 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.
7.2.1.1 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).