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SIP Register Server

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P-2002 Series User's Guide Figure 16 SIP Redirect Server 5.2.3.4 SIP Register Server A SIP register server maintains a database of SIP identity-to-IP address (or domain name) mapping. The register server checks your user name and password when you register. 5.2.4 RTP When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is used to handle voice data transfer. See RFC 1889 for details on RTP. 5.3 NAT NAT (Network Address Translation - NAT, RFC 1631) is the translation of the IP address of a host in a packet. For example, the source address of an outgoing packet, used within one network is changed to a different IP address known within another network. In the simplest form, NAT changes the source IP address of a packet received from a device to another IP address before forwarding the packet towards the destination. When the response comes back, NAT translates the destination address back to the device's IP address and forwards it to the device. NAT routers are commonly used to translate private (or internal) IP addresses in packet headers to public (or external) IP addresses and vice versa. A NAT router maps a private IP address and port pair to a public IP address and port, and whenever the NAT router receives a packet with that public IP address and port, it knows how to reroute the packet back to the private IP address and port. 48 Chapter 5 Introduction to VoIP

Section	Page
P-2002 Series	1
Copyright	3
Federal Communications Commission (FCC) Interference Statement	4
Safety Warnings	6
ZyXEL Limited Warranty	7
Customer Support	8
Table of Contents	11
List of Figures	17
List of Tables	19
Preface	21
Introducing the Prestige	23
1.1 Prestige VoIP Analog Telephone Adaptor Overview	23
1.2 Prestige 2002L VoIP Analog Telephone Adaptor with Lifeline	23
1.3 Features	23
1.4 LEDs	25
1.5 Applications	26
1.5.1 Make Calls via Internet Telephony Service Provider	26
1.5.2 Make Calls via IP-PBX	27
1.5.3 Make Peer-to-peer Calls	27
Introducing the Web Configurator	29
2.1 Web Configurator Overview	29
2.2 Accessing the Prestige Web Configurator	29
2.3 Resetting the Prestige	31
2.3.1 Procedure To Use The Reset Button	31
2.4 Navigating the Prestige Web Configurator	31
2.5 Common Screen Command Buttons	33
System Screens	35
3.1 System Overview	35
3.2 General Setup	35
3.2.1 Domain Name	35
3.3 General Setup Configuration	35
3.4 Configuring Password	36
3.5 Pre-defined NTP Time Servers List	37
3.6 Configuring Time Setting	37
Ethernet Screens	41
4.1 ETHERNET Overview	41
4.1.1 IP Address and Subnet Mask	41
4.1.2 DNS Server Address Assignment	41
4.1.3 PPPoE Encapsulation	42
4.2 Configuring Ethernet	42
4.3 Configuring the Ethernet Management Port	44
Introduction to VoIP	45
5.1 Introduction to VoIP	45
5.2 Introduction to SIP	45
5.2.1 SIP Identities	45
5.2.1.1 SIP Number	45
5.2.1.2 SIP Service Domain	45
5.2.2 SIP Call Progression	46
5.2.3 SIP Servers	46
5.2.3.1 SIP User Agent	46
5.2.3.2 SIP Proxy Server	47
5.2.3.3 SIP Redirect Server	47
5.2.3.4 SIP Register Server	48
5.2.4 RTP	48
5.3 NAT	48
5.3.1 NAT Example	49
5.3.2 NAT Types	49
5.3.2.1 Full Cone NAT	50
5.3.2.2 Restricted Cone NAT	51
5.3.2.3 Port Restricted Cone NAT	52
5.3.2.4 Symmetric NAT	52
5.4 NAT and SIP	53
5.4.1 SIP ALG	53
5.4.2 Use NAT Address on SIP and RTP	53
5.4.3 STUN	54
5.4.4 Outbound Proxy	54
5.5 Pulse Code Modulation	54
5.6 Voice Coding	54
5.6.1 G.711	55
5.6.2 G.729	55
5.7 PSTN Call Setup Signaling	55
5.8 MWI (Message Waiting Indication)	55
VoIP Screens	57
6.1 VoIP Introduction	57
6.2 VoIP Configuration	57
6.3 Custom Tones (IVR)	59
6.3.0.1 Recording Custom Tones	60
6.3.0.2 Listening to Custom Tones	60
6.3.0.3 Deleting Custom Tones	60
6.4 Advanced VoIP Settings Configuration	60
6.5 Quality of Service (QoS)	64
6.5.1 Type Of Service (ToS)	64
6.5.2 DiffServ	64
6.5.2.1 DSCP and Per-Hop Behavior	65
6.5.3 VLAN	65
6.6 QoS Configuration	65
Phone	67
7.1 Phone Introduction	67
7.1.1 Voice Activity Detection/Silence Suppression	67
7.1.2 Comfort Noise Generation	67
7.1.3 Echo Cancellation	67
7.2 Phone Port Configuration	67
7.3 Supplementary Phone Services Overview	69
7.3.1 The Flash Key	69
7.3.2 Europe Type Supplementary Phone Services	69
7.3.2.1 European Call Hold	70
7.3.2.2 European Call Waiting	70
7.3.2.3 European Call Transfer	71
7.3.2.4 European Three-Way Conference	71
7.3.3 USA Type Supplementary Services	71
7.3.3.1 USA Call Hold	71
7.3.3.2 USA Call Waiting	72
7.3.3.3 USA Call Transfer	72
7.3.3.4 USA Three-Way Conference	72
7.4 Common Phone Configuration	72
Phone Book	75
8.1 Phone Book Introduction	75
8.1.1 Speed Dial	75
8.1.1.1 Peer-to-Peer Calls	75
8.1.2 Lifeline (Prestige 2002L)	75
8.2 Speed Dial Configuration	75
8.3 Call Forward	77
8.4 Lifeline Configuration (Prestige 2002L)	80
Logs	81
9.1 Viewing Logs	81
9.2 Log Message Descriptions	82
Maintenance	85
10.1 Maintenance Overview	85
10.2 Status Screen	85
10.3 F/W Upload Screen	86
10.4 Configuration Screen	88
10.4.1 Backup Configuration	89
10.4.2 Restore Configuration	89
10.4.3 Back to Factory Defaults	90
10.5 Restart Screen	91
Phone Usage	93
11.1 Dialing a Telephone Number	93
11.2 Using Speed Dial to Dial a Telephone Number	93
11.3 Internal Calls	93
11.4 Checking the Prestige’s IP Address	93
11.5 Auto Firmware Upgrade	94
Troubleshooting	95
12.1 Problems Starting Up the Prestige	95
12.2 Problems with the LAN or PC LED	95
12.3 Problems with the LAN Interface	96
12.4 Problems with Internet Access	96
12.5 Problems with the Web Configurator	97
12.6 Problems with the Password	97
12.7 Problems with Telephone or Telephone Port	98
12.8 Problems with Voice Service	98
12.9 Pop-up Windows, JavaScripts and Java Permissions	99
12.9.1 Internet Explorer Pop-up Blockers	99
12.9.1.1 Disable Pop-up Blockers	99
12.9.1.2 Enable Pop-up Blockers with Exceptions	100
12.9.2 JavaScripts	102
12.9.3 Java Permissions	104
12.9.3.1 JAVA (Sun)	105
Product Specifications	107
Setting up Your Computer’s IP Address	111
IP Subnetting	123
PPPoE	131
Internal SPTGEN	133
Index	161
Numerics	161
A	161
B	161
C	161
D	162
E	162
F	162
G	163
H	163
I	163
J	163
K	163
L	163
M	164
N	164
O	164
P	164
Q	165
R	165
S	166
T	167
U	167
V	167
W	167

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P-2002 Series User’s Guide

Chapter 5 Introduction to VoIP

Figure 16

SIP Redirect Server

5.2.3.4

SIP Register Server

A SIP register server maintains a database of SIP identity-to-IP address (or domain name)

mapping. The register server checks your user name and password when you register.

5.2.4

RTP

When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is used to

handle voice data transfer. See RFC 1889 for details on RTP.

5.3

NAT

NAT (Network Address Translation - NAT, RFC 1631) is the translation of the IP address of a

host in a packet. For example, the source address of an outgoing packet, used within one

network is changed to a different IP address known within another network.

In the simplest form, NAT changes the source IP address of a packet received from a device to

another IP address before forwarding the packet towards the destination. When the response

comes back, NAT translates the destination address back to the device’s IP address and

forwards it to the device.

NAT routers are commonly used to translate private (or internal) IP addresses in packet

headers to public (or external) IP addresses and vice versa. A NAT router maps a private IP

address and port pair to a public IP address and port, and whenever the NAT router receives a

packet with that public IP address and port, it knows how to reroute the packet back to the

private IP address and port.