Billion BiPAC-7401VGP Router Manual PDF (Setup & Configuration Guide)

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VoIP/(802.11g) ADSL2+ Router

Chapter 4: Configuration

General Settings

This section reflects and contains basic settings for the VoIP module from selected

provider in the Wizard section. Fail to provide correct information will halt making calls out

to the Internet.

SIP Device Parameters

SIP:

To use SIP as VoIP call signaling protocol.

Default is set to

Disable.

Silence Suppression (VAD):

Voice Activation Detection prevents transmitting the nature

silence to consume the bandwidth. It is also known as Silence Suppression which is a

software application that ensures the bandwidth is reserved only when voice activity is

activated.

Default is set to

Enable.

Echo Cancellation:

G.168 echo canceller is an ITU-T standard.

It is used for isolating the

echo while you are on the phone. This helps you not to hear much of your own voice

reflecting on the phone while you talk. Default is set to

Enable.

RTP Port:

Provide the based value from the media (RTP) ports that are assigned for various

endpoints and the different call sessions that may exist within an end-point. (Range from 5100 to

65535, default value is 5100)

Region:

This selection is a drop-down box, which allows user to select the country for which the

VoIP device must work. When a country is selected, the country parameters are automatically

loaded.

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VoIP/(802.11g) ADSL2+ Router

Chapter 4: Configuration

Voice QoS, DSCP Marking:

Differentiated Services Code Point (DSCP), it is the first 6 bits in

the ToS byte. DSCP Marking allows users to classify traffic based on DSCP value and send

packets to next Router.

Setting for Phone Port 1

Registrar Address(or Hostname):

Indicate the SIP registrar IP address.

Registrar Port:

Specify the port of the SIP registrar on which it will listen for register requests

from VoIP device.

Expire:

Expire time for the registration message sending.

User Domain/Realm:

Set different domain name for the SIP proxy server.

Outbound Proxy Address:

Indicate the SIP outbound proxy server IP address. This parameter

is very useful when VoIP device is behind a NAT.

Outbound Proxy Port:

Specify the port of the SIP outbound proxy on which it will listen for

messages.

How to register to SIP Server

1) In Wizard Section, select your SIP Service Provider and input information in the filed of

Phone Number, Authentication Username

and

Authentication Password.

2) In Wizard Section, click Apply

button to apply the settings.

3) In General Settings, make sure general SIP information are correctly inserted.

4) In General Settings, click Apply

button to apply the settings.

5) In General Settings, click Synch Now

button to register the account(s) with your SIP

server.

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VoIP/(802.11g) ADSL2+ Router

Chapter 4: Configuration

Phone Ports

This section displays status and allows you to edit the account information of your Phones.

Click

Edit

to update your phone information.

Login Account Configuration

Phone Number:

This parameter holds the registration ID of the user within the SIP registrar.

Authentication Username:

Same as Phone Number.

Authentication Password:

This parameter holds the password used for authentication within SIP

registrar.

Confirm Password:

Re-enter the password for confirmation.

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VoIP/(802.11g) ADSL2+ Router

Chapter 4: Configuration

Display Name:

This parameter will be appeared on the Caller ID.

Codec Preference

Codec is known as Coder-Decoder used for data signal conversion.

Set the priority of voice

compression; Priority 1 owns the top priority.

G.729:

It is used to encoder and decoder voice information into a single packet which reduces the

bandwidth consumption.

8kbps bandwidth is needed.

G.71

1

μ-LAW:

It is a basic non-compressed encoder and decoder technique. μ-LAW uses pulse

code modulation (PCM) encoder and decoder to convert 14-bit linear sample.

64kbps bandwidth

is needed.

G.711A-LAW:

It is a basic non-compressed encoder and decoder technique. μ-LAW uses pulse

code modulation (PCM) encoder and decoder to convert 13-bit linear sample.

64kbps bandwidth

is needed.

Non

-

used:

Only available in Priority 2 and 3.

It is selected if codec is not to be used.

Note:

Codec priority is assigned in the order as

G.729 > G.71

1

μ-LAW > G.711A-LAW

Speed Dial

It is for you to store frequently used telephone numbers which you can press number from 0 to 9 and the

pound sign (#) to activate this function. For example, speed dial to phone number lists on 9, just press

9

then

#

.

Your router will automatically call out to number listed on entry 9.

Indicate remote user’s IP address or domain name if this remote user does not register in the SIP server.

If remote user is registered in the SIP server, this field is related to the SIP server’s IP / Domain name.

For examples:

If your friend Tim gives you a SIP URL as sip: [email protected] then you can fill in as

[email protected]

.

If your friend Felix gives you a SIP URL as sip: [email protected] then you can fill in as

[email protected]

.

If your friend Greg gives you an IP address "192.246.56.56" only, then you can fill in as “192.246.56.56”.

In case, some of users may use DDNS, you can fill in with domain name as well.

Volume Control

Volume control helps you to adjust the voice quality of telephone to the best comfortable listening level.

Press “

-

“, the minus sign, to reduce either microphone or/both speaker’s level of your telephone.

Press “

+

“, the plus sign, to increase either microphone or/both speaker’s level of your telephone.

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VoIP/(802.11g) ADSL2+ Router

Chapter 4: Configuration

PSTN Dial Plan

This section enables you to configure “VoIP with PSTN switching” on your system. You

can define a range of dial plans to make regular call from VoIP switching to PSTN line.

Prefix numbers is essential key to make a distinguishing between VoIP and Regular

phone call. If actual numbers dialed matches with prefix number defined in this dial plan,

the dialed number will be routed to the PSTN to make a regular call. Otherwise, the

number will be routed to the VoIP networks.

Reminder!

In order to utilize this feature, you must have registered and connected to your

SIP Server fist.

Prefix:

Specify number(s) for switching to a PSTN call.

Number of Digits:

Specify the total number of digits wish to dial out. Maximum digit

number is 15.

Action:

Specify a dialing method you wish to make PSTN call(s).

~

Dial with Prefix:

The dialed number

with

prefix will be sent call through the PSTN.

NOTE:

The actual dialed number of valid digits length

requires

matching in the

Number of Digits

filed.

~

Dial without Prefix:

The dialed number will be sent call through the PSTN

without

prefix.

NOTE:

The actual dialed number of valid digits length

requires

matching in the

Number of Digits

filed.

~

Dial at Timeout:

The dialed number will be sent call through the PSTN

with

the prefix when

timeout starts. This timeout activates when no more digits are dialed in a specific duration.

NOTE:

The actual dialed number of valid digits length

MUST NOT

exceed in the

Number of Digits

filed.

~

Dial at Timeout no Prefix:

The dialed number will be sent call through the PSTN

without

prefix when timeout starts. This timeout activates when no more digits are dialed in a specific

duration.

NOTE:

The actual dialed number of valid digits length

MUST NOT

exceed in the

Number of Digits

filed.

79

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