Page 191 / 344 Scroll up to view Page 186 - 190
Chapter 16 VoIP
P-2601HN(L)-F1 Series User’s Guide
191
Each field is described in the following table.
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.
Table 55
VoIP > FXO
LABEL
DESCRIPTION
Pre-Fix For FXO Outgoing Call
Pre-Fix Number
Enter 1 - 7 numbers you dial before you dial the phone number, if you
want to make a regular phone call while one of your SIP accounts is
registered. These numbers tell the ZyXEL Device that you want to make
a regular phone call.
Voice Features
Active G.168
Select this if you want to eliminate the echo caused by the sound of
your voice reverberating in the telephone receiver while you talk.
Active VAD
Select this if the ZyXEL Device should stop transmitting when you are
not speaking. This reduces the bandwidth the ZyXEL Device uses.
SIP Fail Over
Force to SIP if
PSTN un-
plugged
Select this check box to have the ZyXEL Device redirect outgoing calls
to the registered SIP account if the ZyXEL Device is not connected to
the PSTN network.
When you try to make a PSTN call, but the PSTN port on the ZyXEL
Device is unplugged, the ZyXEL Device uses the phone port’s registered
SIP account to make the call.
Apply
Click
Apply
to save your changes.
Cancel
Click
Cancel
to restore your previously saved settings.
Page 192 / 344
Chapter 16 VoIP
P-2601HN(L)-F1 Series User’s Guide
192
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.
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.
Page 193 / 344
Chapter 16 VoIP
P-2601HN(L)-F1 Series User’s Guide
193
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").
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 90
SIP User Agent
A
B
Page 194 / 344
Chapter 16 VoIP
P-2601HN(L)-F1 Series User’s Guide
194
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
.
2
The SIP proxy server forwards the call invitation to
C
.
Figure 91
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
Page 195 / 344
Chapter 16 VoIP
P-2601HN(L)-F1 Series User’s Guide
195
3
Client device
A
then sends the call invitation to client device
C
.
Figure 92
SIP Redirect Server
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.
RTP
When you make a VoIP call using SIP, the RTP (Real time Transport Protocol) is
used to handle voice data transfer. See RFC 3550 for details on RTP.
Pulse Code Modulation
Pulse Code Modulation (PCM) measures analog signal amplitudes at regular time
intervals and converts them into bits.
SIP Call Progression
The following figure displays the basic steps in the setup and tear down of a SIP
call. A calls B.
Table 56
SIP Call Progression
A
B
1. INVITE
2. Ringing
3. OK
4. ACK
1
2
3
A
B
C

Rate

4 / 5 based on 1 vote.

Bookmark Our Site

Press Ctrl + D to add this site to your favorites!

Share
Top