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For viewing each trusted CA certificate, click
View
to open the certificate detail information
window. If you want to delete a CA certificate, choose the one and click
Delete
to remove all
the certificate information.
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3.9 VoIP
Voice over IP network (VoIP) enables you to use your broadband Internet connection to
make toll quality voice calls over the Internet.
There are many different call signaling protocols, methods by which VoIP devices can talk
to each other. The most popular protocols are SIP, MGCP, Megaco and H.323. These
protocols are not all compatible with each other (except via a soft-switch server).
The Vigor V models support the SIP protocol as this is an ideal and convenient deployment
for the ITSP (Internet Telephony Service Provider) and softphone and is widely supported.
SIP is an end-to-end, signaling protocol that establishes user presence and mobility in VoIP
structure. Every one who wants to talk using his/her SIP Uniform Resource Identifier, “SIP
Address”. The standard format of SIP URI is
sip: user:password @ host: port
Some fields may be optional in different use. In general, "host” refers to a domain. The
“userinfo” includes the user field, the password field and the @ sign following them. This is
very similar to a URL so some may call it “SIP URL”. SIP supports peer-to-peer direct
calling and also calling via a SIP proxy server (a role similar to the gatekeeper in H.323
networks), while the MGCP protocol uses client-server architecture, the calling scenario
being very similar to the current PSTN network.
After a call is setup, the voice streams transmit via RTP (Real-Time Transport Protocol).
Different codecs (methods to compress and encode the voice) can be embedded into RTP
packets. Vigor V models provide various codecs, including G.711 A/μ-law, G.723, G.726
and G.729 A & B. Each codec uses a different bandwidth and hence provides different levels
of voice quality. The more bandwidth a codec uses the better the voice quality, however the
codec used must be appropriate for your Internet bandwidth.
Usually there will be two types of calling scenario, as illustrated below:
z
Calling via SIP Servers
First, the Vigor V models of yours will have to register to a SIP Registrar by sending
registration messages to validate. Then, both parties’ SIP proxies will forward the
sequence of messages to caller to establish the session.
If you both register to the same SIP Registrar, then it will be illustrated as below:
The major benefit of this mode is that you don’t have to memorize your friend’s IP
address, which might change very frequently if it’s dynamic. Instead of that, you will
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only have to using
dial plan
or directly dial your friend’s
account name
if you are
with the same SIP Registrar. Please refer to the
Example 1 and 2 in the Calling
Scenario.
z
Peer-to-Peer
Before calling, you have to know your friend’s IP Address. The Vigor VoIP Routers
will build connection between each other. Please refer to the
Example 3 in the Calling
Scenario.
Our Vigor V models firstly apply efficient codecs designed to make the best use of
available bandwidth, but Vigor V models also equip with automatic QoS assurance.
QoS Assurance assists to assign high priority to voice traffic via Internet. You will
always have the required inbound and outbound bandwidth that is prioritized
exclusively for Voice traffic over Internet but you just get your data a little slower and
it is tolerable for data traffic.
Concepts of ISDN On-Net and Off-Net (for 2800VGi only)
Different directions generate different meanings for on-net and off-net.
y
ISDN On-Net
means a user (A) calls another one (B) through ISDN line, routers and
Internet with the yellow route displayed in the following graphic.
User A connects to
S0 extern
of switchboard through ISDN line. The switchboard
connects to a router(C) with
ISDN cable
. The router connects to Internet through
WAN port with ADSL cable. When A calls B, the voice signal will be sent to router (C)
through ISDN interface. Next, the voice signal will be transferred to B through Internet.
Such route is named
ISDN On-Net
(see yellow route).
y
ISDN Off-Net
means a user (B) calls another one (A) through Internet, router and
ISDN line with the blue route displayed in the following graphic.
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User B connects a router with FXS port and accesses Internet through WAN port of
that router. When B calls A, the voice signal will be sent to a remote router (C) through
Internet. Then, the voice signal will be passed to switchboard (via
S0 intern
) and
transferred to A through ISDN line (leaving Internet). Such route is named
ISDN
Off-Net
(see blue route).
Below shows the menu items for Certificate Management.
3.9.1 DialPlan
This page allows you to set phone book and digit map for the VoIP function. Click the
Phone Book
and
Digit Map
links on the page to access into next pages for dialplan settings.
Phone Book
In this section, you can set your VoIP contacts in the “phonebook”. It can help you to make
calls quickly and easily by using “speed-dial”
Phone Number
. There are total 60 index
entries in the phonebook for you to store all your friends and family members’ SIP
addresses.
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Click any index number to display the dial plan setup page.
Enable
Click this to enable this entry.
Phone Number
The speed-dial number of this index. This can be any number
you choose, using digits
0-9
and
*
.
Display Name
The Caller-ID that you want to be displayed on your friend’s
screen. This let your friend can easily know who’s calling
without memorizing lots of SIP URL Address.
SIP URL
Enter your friend’s SIP Address
Digit Map
For the convenience of user, this page allows users to edit prefix number for the SIP account
with adding number, stripping number or replacing number. It
is used to help user having a quick and easy way to dial out
through VoIP interface.

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