Home » ZyXEL Manuals » Networking » ZyXEL V630 » Manual Viewer

ZyXEL V630 User Guide - Page 131

Voice Coding

Get ZyXEL V630 PDF manuals and user guides

Add to My Manuals
Save this manual to your list of manuals

Page 131 highlights

Chapter 14 SIP Account Setup 14.1.6.1 STUN STUN (Simple Traversal of User Datagram Protocol (UDP) through Network Address Translators) allows the V630 to find the presence and types of NAT routers and/or firewalls between it and the public Internet. STUN also allows the V630 to find the public IP address that NAT assigned, so the V630 can embed it in the SIP data stream. STUN does not work with symmetric NAT routers or firewalls. See RFC 3489 for details on STUN. The following figure shows how STUN works. 1 The V630 (A) sends SIP packets to the STUN server (B). 2 The STUN server (B) finds the public IP address and port number that the NAT router used on the V630's SIP packets and sends them to the V630. 3 The V630 uses the public IP address and port number in the SIP packets that it sends to the SIP server (C). Figure 131 STUN 14.1.6.2 Outbound Proxy Your VoIP service provider may host a SIP outbound proxy server to handle all of the V630's VoIP traffic. This allows the V630 to work with any type of NAT router and eliminates the need for STUN or a SIP ALG. Turn off a SIP ALG on a NAT router in front of the V630 to keep it from retranslating the IP address (since this is already handled by the outbound proxy server). 14.1.7 Voice Coding A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital signals back into voice signals. The V630 supports the following codecs. • G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal amplitudes at regular time intervals (sampling) and converts them into digital bits (quantization). Quantization "reads" the analog signal and then "writes" it to the nearest digital value. For this reason, a digital sample is usually slightly different from its analog original (this difference is known as "quantization noise"). • G.726 is an ADPCM waveform codec that uses a lower bit rate than standard PCM conversion. G.726 operates at 16, 24, 32 or 40 kbps. V630 User's Guide 131

Section	Page
User’s Guide	1
About This User's Guide	3
Document Conventions	4
Safety Warnings	6
Contents Overview	9
Table of Contents	11
List of Figures	17
List of Tables	23
Introduction	27
Introducing the V630	29
1.1 Overview	29
1.1.1 Making Calls via Internet Telephony Service Provider	29
1.1.2 Making Calls via IP-PBX	30
1.1.3 Making Peer to Peer Calls	30
1.2 Ways to Manage the V630	31
1.3 Good Habits for Managing the V630	31
LCD Screen Menus	33
Using the LCD Screen	35
2.1 Entering the Menu System	35
2.2 Navigation	35
2.3 Entering Numbers, Letters and Symbols	36
2.4 LCD Menu Overview	37
Call Log LCD Menus	41
3.1 Call Log	41
3.2 Received Calls	41
Profiles LCD Menus	43
4.1 Profiles Setup	43
4.2 Phone Profile Options	43
4.3 Phone Profile Personalization	44
4.4 Tone Personalization	45
4.5 Ring Tone Personalization	45
4.6 Volume Personalization	46
4.7 Ring Volume Personalization	46
4.8 Ring Mode Personalization	47
4.9 Adding a Phone Profile	48
General Setup LCD Menus	49
5.1 General Setup	49
5.2 Date and Time Setup	49
5.3 Manual Date or Time Setup	50
5.4 Manual Time Setup	51
5.5 Manual Date Setup	51
5.6 Using a Time Server	52
5.7 Specifying a Time Server	52
5.8 Time Zone Setup	53
5.9 General Phone Setup	53
5.10 Language Setup	54
5.11 Keypad Lock Setup	55
5.12 Backlight Setup	55
5.13 Quick Button Setup	56
5.14 Up Quick Button Setup	57
5.15 Enabling or Disabling the Web Configurator	57
5.16 Firmware Upgrade from an HTTP Server	58
5.17 HTTP Firmware Upgrade Server Address	59
5.18 Restore Factory Default Settings	59
5.19 Call Settings	60
5.20 Call Forwarding	60
5.21 Call Forwarding Number	61
5.22 Call Forwarding Type	61
5.23 Call Forwarding No Answer Time	62
5.24 Call Forwarding No Answer Time	63
5.25 Send Caller ID	63
5.26 Information	64
5.27 TCP/IP Information	64
5.28 WLAN Information	65
5.29 SIP Information	66
5.30 Hardware Information	66
5.31 Log Information	67
Network LCD Menus	69
6.1 Network Setup	69
6.2 Site Scan	69
6.3 Wireless Security	70
6.4 WLAN Profiles	71
6.5 WLAN Profiles List	71
6.6 WLAN Profile	72
6.7 Adding a WLAN Profile	72
6.8 Setting the SSID	73
6.9 Setting the Wireless Security Type	73
6.10 Setting the Wireless Security Key	74
6.11 IP Settings	75
6.12 Static IP Settings	76
6.13 Static IP Address Setup	76
6.14 PPPoE Settings	77
6.15 PPPoE Username	77
6.16 Selecting the SIP Account	78
6.17 SIP Profiles	79
6.18 SIP Profiles List	79
6.19 SIP Profile	80
6.20 Adding a SIP Profile	80
6.21 Editing the New SIP Profile	81
6.22 SIP Display Name	82
6.23 SIP Phone Number	82
6.24 SIP Server	83
6.25 SIP Server Address	83
6.26 SIP Port Number	84
6.27 SIP Proxy Setup	85
6.28 SIP Proxy Address	85
6.29 SIP Proxy Port	86
6.30 SIP Proxy User Name	86
6.31 SIP Proxy Password	87
6.32 NAT Traversal	88
6.33 STUN Setup	88
6.34 STUN Server Address	89
6.35 STUN Port Number	90
6.36 Outbound Proxy Setup	90
6.37 Outbound Proxy Server Address	91
6.38 Outbound Proxy Port Number	91
6.39 NAT Keep Alive Time	92
6.40 SIP Server Expire Time	93
6.41 Codec Order	93
6.42 Ping Test	94
6.43 Manual Ping Test	94
6.44 Ping Test in Progress	95
6.45 Reconnect	95
The Phonebook	97
7.1 Opening the Phonebook	97
7.2 Adding a Phonebook Entry	97
7.3 Selecting a Phonebook Entry	99
7.4 Calling a Phonebook Contact	100
7.5 Calling a Number Not in the Phonebook	100
7.6 Checking a Contact’s Details	100
7.7 Editing a Phonebook Entry	101
7.8 Deleting a Phonebook Entry	102
7.9 Contact Groups	102
7.10 Editing a Contact Group’s Members	103
7.11 Editing a Contact Group’s Ring Tone	104
7.12 Speed Dial	104
7.13 Adding a Speed Dial Entry	104
7.14 Editing a Speed Dial Entry	105
7.15 Deleting All Phonebook Entries	105
7.16 Phonebook Storage Space	106
Call Options	107
8.1 Call Volume	107
8.2 Call Options	107
The Web Configurator	109
Introducing the Web Configurator	111
9.1 Web Configurator Overview	111
9.2 Accessing the Web Configurator	111
9.2.1 Navigation Panel	113
9.2.2 Main Window	114
Information Screen	115
10.1 Information Screen	115
WLAN Profile	117
11.1 Wireless Network Overview	117
11.2 Wireless Security Overview	118
11.2.1 SSID	118
11.2.2 User Authentication	119
11.2.3 Encryption	119
11.3 IP Address Assignment	119
11.3.1 DHCP Client	119
11.3.2 Static IP	119
11.3.3 PPPoE	119
11.4 DNS Server	119
11.5 WLAN Profile Screen	120
Call Setting	123
12.1 Call Setting Screen	123
Phone Book	125
13.1 Phone Book Screen	125
13.1.1 Phone Book Add or Edit Screen	126
SIP Account Setup	127
14.1 Introduction to VoIP	127
14.1.1 Introduction to SIP	127
14.1.2 SIP Identities	127
14.1.3 SIP Call Progression	128
14.1.4 SIP Client Server	128
14.1.5 RTP	130
14.1.6 NAT and SIP	130
14.1.7 Voice Coding	131
14.2 SIP Settings Screen	132
Auto Provision	135
15.1 Auto Provision Screen	135
System, Troubleshooting, and Specifications	137
System	139
16.1 Password Screen	139
Troubleshooting	141
17.1 Power, Hardware Connections, and LEDs	141
17.2 V630 Web Configurator Access and Login	142
17.3 Wireless LAN	143
17.4 Phone Calls	144
Product Specifications	147
Appendices and Index	151
Setting Up Your Computer’s IP Address	153
Wireless LANs	177
Pop-up Windows, JavaScripts and Java Permissions	189
IP Addresses and Subnetting	197
Legal Information	207
Customer Support	211

Match case Limit results 1 per page

Chapter 14 SIP Account Setup

V630 User’s Guide

131

14.1.6.1

STUN

STUN (Simple Traversal of User Datagram Protocol (UDP) through Network Address

Translators) allows the V630 to find the presence and types of NAT routers and/or firewalls

between it and the public Internet. STUN also allows the V630 to find the public IP address

that NAT assigned, so the V630 can embed it in the SIP data stream. STUN does not work

with symmetric NAT routers or firewalls. See RFC 3489 for details on STUN.

The following figure shows how STUN works.

The V630 (

) sends SIP packets to the STUN server (

The STUN server (

) finds the public IP address and port number that the NAT router

used on the V630’s SIP packets and sends them to the V630.

The V630 uses the public IP address and port number in the SIP packets that it sends to

the SIP server (

Figure 131

STUN

14.1.6.2

Outbound Proxy

Your VoIP service provider may host a SIP outbound proxy server to handle all of the V630’s

VoIP traffic. This allows the V630 to work with any type of NAT router and eliminates the

need for STUN or a SIP ALG. Turn off a SIP ALG on a NAT router in front of the V630 to

keep it from retranslating the IP address (since this is already handled by the outbound proxy

server).

14.1.7

Voice Coding

A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital

signals back into voice signals. The V630 supports the following codecs.

•

G.711

is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal

amplitudes at regular time intervals (sampling) and converts them into digital bits

(quantization). Quantization “reads” the analog signal and then “writes” it to the nearest

digital value. For this reason, a digital sample is usually slightly different from its analog

original (this difference is known as “quantization noise”).

•

G.726

is an ADPCM waveform codec that uses a lower bit rate than standard PCM

conversion. G.726 operates at 16, 24, 32 or 40 kbps.