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Page 246 / 424 Scroll up to view Page 241 - 245
Chapter 16 VoIP
P-2612HNU-Fx User’s Guide
246
To access this screen, click
VoIP > Phone > Call Rule
.
Figure 103
VoIP > Phone > Call Rule
Each field is described in the following table.
Table 64
VoIP > Phone > Call Rule
LABEL
DESCRIPTION
Speed Dial
Use this section to create or edit speed-dial entries.
#
Select the speed-dial number you want to use for this phone number.
Number
Enter the SIP number you want the ZyXEL Device to call when you dial
the speed-dial number.
Description
Enter a short description to identify the party you call when you dial the
speed-dial number. You can use up to 127 printable ASCII characters.
Add
Click this to use the information in the
Speed Dial
section to update
the
Speed Dial Phone Book
section.
Phone Book
Use this section to look at all the speed-dial entries and to erase them.
#
This field displays the speed-dial number you should dial to use this
entry.
Number
This field displays the SIP number the ZyXEL Device calls when you dial
the speed-dial number.
Description
This field displays a short description of the party you call when you dial
the speed-dial number.
Modify
Use this field to edit or erase the speed-dial entry.
Click the
Edit
icon to copy the information for this speed-dial entry into
the
Speed Dial
section, where you can change it. Click
Add
when you
finish editing to change the configurations.
Click the
Delete
icon to erase this speed-dial entry.
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Chapter 16 VoIP
P-2612HNU-Fx User’s Guide
247
16.9
Technical Reference
This section contains background material relevant to the
VoIP
screens.
16.9.1
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.
16.9.2
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.
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.
Clear
Click this to erase all the speed-dial entries.
Cancel
Click this to set every field in this screen to its last-saved value.
Table 64
VoIP > Phone > Call Rule (continued)
LABEL
DESCRIPTION
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Chapter 16 VoIP
P-2612HNU-Fx User’s Guide
248
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 ([email protected] for
example) or numbers like a telephone number ([email protected]
for example).
SIP Service Domain
The SIP service domain of the VoIP service provider is the domain name in a SIP
URI. For example, if the SIP address is
, then
“VoIP-provider.com” is the SIP service domain.
SIP Registration
Each ZyXEL Device is an individual SIP User Agent (UA). To provide voice service,
it has a public IP address for SIP and RTP protocols to communicate with other
servers.
A SIP user agent has to register with the SIP registrar and must provide
information about the users it represents, as well as its current IP address (for the
routing of incoming SIP requests). After successful registration, the SIP server
knows that the users (identified by their dedicated SIP URIs) are represented by
the UA, and knows the IP address to which the SIP requests and responses should
be sent.
Registration is initiated by the User Agent Client (UAC) running in the VoIP
gateway (the ZyXEL Device). The gateway must be configured with information
letting it know where to send the REGISTER message, as well as the relevant user
and authorization data.
A SIP registration has a limited lifespan. The User Agent Client must renew its
registration within this lifespan. If it does not do so, the registration data will be
deleted from the SIP registrar's database and the connection broken.
The ZyXEL Device attempts to register all enabled subscriber ports when it is
switched on. When you enable a subscriber port that was previously disabled, the
ZyXEL Device attempts to register the port immediately.
Authorization Requirements
SIP registrations (and subsequent SIP requests) require a username and
password for authorization. These credentials are validated via a challenge /
response system using the HTTP digest mechanism (as detailed in RFC3261, "SIP:
Session Initiation Protocol").
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Chapter 16 VoIP
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249
SIP Servers
SIP is a client-server protocol. A SIP client is an application program or device that
sends SIP requests. A SIP server responds to the SIP requests.
When you use SIP to make a VoIP call, it originates at a client and terminates at a
server. A SIP client could be a computer or a SIP phone. One device can act as
both a SIP client and a SIP server.
SIP User Agent
A SIP user agent can make and receive VoIP telephone calls. This means that SIP
can be used for peer-to-peer communications even though it is a client-server
protocol. In the following figure, either
A
or
B
can act as a SIP user agent client to
initiate a call.
A
and
B
can also both act as a SIP user agent to receive the call.
Figure 104
SIP User Agent
SIP Proxy Server
A SIP proxy server receives requests from clients and forwards them to another
server.
In the following example, you want to use client device
A
to call someone who is
using client device
C
.
1
The client device (
A
in the figure) sends a call invitation to the SIP proxy server
B
.
A
B
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Chapter 16 VoIP
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250
2
The SIP proxy server forwards the call invitation to
C
.
Figure 105
SIP Proxy Server
SIP Redirect Server
A SIP redirect server accepts SIP requests, translates the destination address to
an IP address and sends the translated IP address back to the device that sent the
request. Then the client device that originally sent the request can send requests
to the IP address that it received back from the redirect server. Redirect servers
do not initiate SIP requests.
In the following example, you want to use client device
A
to call someone who is
using client device
C
.
1
Client device
A
sends a call invitation for
C
to the SIP redirect server
B
.
2
The SIP redirect server sends the invitation back to
A
with
C
’s IP address (or
domain name).
B
A
C
1
2

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