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54
UPnP is available on Windows XP
and the router provide the associated support for MSN
Messenger to allow full use of the voice, video and messaging features.
Enable UPNP Service
Accordingly, you can enable either the
Connection Control
Service
or
Connection Status Service
.
After setting
Enable UPNP Service
setting, an icon of
IP Broadband Connection on
Router
on Windows XP/Network Connections will appear. The connection status and
control status will be able to be activated. The NAT Traversal of UPnP enables the
multimedia features of your applications to operate. This has to manually set up port
mappings or use other similar methods. The screenshots below show examples of this
facility.
The UPnP facility on the router enables UPnP aware applications such as MSN Messenger to
discover what are behind a NAT router. The application will also learn the external IP
address and configure port mappings on the router. Subsequently, such a facility forwards
packets from the external ports of the router to the internal ports used by the application.
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The reminder as regards concern about Firewall and UPnP
Can't work with Firewall Software
Enabling firewall applications on your PC may cause the UPnP function not working
properly. This is because these applications will block the accessing ability of some
network ports.
Security Considerations
Activating the UPnP function on your network may incur some security threats. You
should consider carefully these risks before activating the UPnP function.
¾
Some Microsoft operating systems have found out the UPnP weaknesses and hence
you need to ensure that you have applied the latest service packs and patches.
¾
Non-privileged users can control some router functions, including removing and
adding port mappings.
The UPnP function dynamically adds port mappings on behalf of some UPnP-aware
applications. When the applications terminate abnormally, these mappings may not be
removed.
3.5 VoIP
Note:
This function is used for “V” models.
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
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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
only have to using
dial plan
or directly dial your friend’s
account name
if you are
with the same SIP Registrar.
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.
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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.
3.5.1 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.
Loop through
and
Backup Phone Number
will be displayed if you are using Vigor
2710Vn/2710VDn for setting the phone book.
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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.
Dial Out Account
Choose one of the SIP accounts for this profile to dial out. It is
useful for both sides (caller and callee) that registered to
different SIP Registrar servers. If caller and callee do not use

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