Billion 800VGT Router Manual PDF (Setup & Configuration Guide)

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Billion 800VGT Router

Profile

Name:

A

user-defined

name

to

identify

the

profile.

Registrar

Address(or

Hostname):

Indicate

the

VoIP

SIP

registrar’s

IP

address.

Registrar

Port:

Specify

the

port

on

which

the

VoIP

SIP

registrar

will

listen

for

register

requests

from

VoIP

devices.

Expire:

Expire

time

for

the

registration

message

sending.

User

Domain/Realm:

Set

different

domain

names

for

the

VoIP

SIP

proxy

server.

Outbound

Proxy

Address:

Indicate

the

VoIP

SIP

outbound

proxy

server

IP

address.

This

parameter

is

very

useful

when

your

VoIP

device

is

behind

NAT.

Outbound

Proxy

Port:

Specify

the

port

on

which

the

VoIP

SIP

outbound

proxy

will

listen

for

messages.

Phone

Number:

This

is

the

registration

ID

of

the

user

as

listed

in

the

VoIP

SIP

registrar.

Authentication

Username:

Same

as

Phone

Number.

Authentication

Password:

This

is

the

password

used

for

authentication

to

the

VoIP

SIP

registrar.

Confirm

Password:

Re-enter

the

password

for

confirmation.

Display

Name:

This

is

what

will

be

displayed

on

a

Caller

ID

system.

General

Settings

This

section

contains

the

basic

settings

for

the

VoIP

module

from

the

selected

provider

in

the

Wizard

section.

If

you

do

not

provide

correct

information

here,

you

will

be

unable

to

make

calls

over

the

Internet.

1

01

Chapter

4:

Configuration

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Billion 800VGT Router

SIP

Device Parameters

SIP:

Select

whether

you

wish

to

use

SIP

as

VoIP

call

signaling

protocol.

The

default

setting

is

Disable.

Silence

Suppression

(VAD):

Voice

Activation

Detection

(VAD)

prevents

the

transmission

of

silence

since

it

will

unnecessarily

consume

your

bandwidth.

This

function

is

also

known

as

Silence

Suppression,

and

it

is

a

software

application

that

ensures

the

bandwidth

is

used

only

when

voice

activity

is

activated.

The

default

setting

is

Enable.

Echo

Cancellation:

G.168

echo

cancellation

is

an

ITU-T

standard.

It

is

used

for

removing

the

echo

while

you

are

on

the

phone.

If

it

is

enabled,

this

will

mean

that

you

will

not

hear

too

much

of

your

own

voice

on

the

phone

while

you

talk.

The

default

setting

is

Enable.

RTP

Port:

Provide

the

base

value

for

the

media

(RTP)

ports.

These

ports

are

assigned

to

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.

Voice

QoS,

:

Differentiated

Services

Code

Point

(DSCP),

it

is

the

first

6

bits

in

the

ToS

byte.

DSCP

Marking

allows

users

to

assign

specific

application

traffic

to

be

executed

in

priority

by

backbone

Routers,

based

on

the

DSCP

value.

See

Table

4.

The

DSCP

Mapping

Table:

Note:

To

be

sure

that

all

the

router(s)

through-out

the

QoS

network

backbone

have

the

capability

of

executing

and

checking

the

DSCP..

Setting for Phone Port 1

Registrar

Address(or

Hostname):

Indicate

the

SIP

registrar

IP

address.

Registrar

Port:

Specify

the

port

on

which

the

SIP

registrar

listens

for

register

requests

from

VoIP

devices.

Expire:

Expiry

time

for

registration

message

sending.

User

Domain/Realm:

Set

a

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

your

VoIP

device

is

behind

NAT.

Outbound

Proxy

Port:

Specify

the

port

on

which

the

VoIP

SIP

outbound

proxy

will

listen

for

messages.

Setting for Phone Port 2

Please

refer

to

the

descriptions

in

“Setting

for

Phone

Port

1”.

How to register

with a

SIP

Server

1)

On

the

Wizard

Section

page,

select

your

VoIP

Service

Provider

and

provide

information

in

the

following

fields:

Phone

Number,

Authentication

Username

and

Authentication

Password.

2)

On

the

Wizard

Section

page,

click

Apply

to

apply

the

settings.

3)

On

the

General

Settings

page,

make

sure

the

general

VoIP

SIP

information

is

correctly

inserted.

4)

On

the

General

Settings

page,

click

Apply

to

apply

the

settings.

5)

On

the

General

Settings

page,

click

Synch Now

to

register

the

account(s)

with

your

VoIP

server.

1

02

Chapter

4:

Configuration

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Billion 800VGT Router

Advanced – Parameters

VoIP

through

IP

Interface:

IP

Interface

decides

where

to

send/receive

the

VoIP

traffic.

Options

include:

ipwan

and

iplan.

An

easy

way

to

select

the

interface

option

is

to

check

the

location

of

the

SIP

server.

If

it

is

located

somewhere

on

the

Internet,

then

select

ipwan.

If

the

VoIP

SIP

server

is

on

the

local

network

then

select

iplan.

Voice

Frame

Size:

Voice

Frame

size

can

be

set

anywhere

between

10ms

and

60ms.

The

function

of

Voice

Frame

Size

is

how

many

milliseconds

the

Voice

packets

will

be

queued

for,

before

being

sent

out.

Billion 800VGT

Router

The

ideal

setting

is

to

have

the

same

frame

size

for

both

Caller

and

Receiver.

PSTN

Auto-fallback:

Whenever

VoIP

SIP

response

is

an

error

code

that

matches

the

codes

in

the

Edit

section,

the

VoiP

calls

will

automatically

fall

back

to

a

PSTN.

Click

Edit

to

add

or

remove

codes.

Be

sure

that

the

codes

are

separated

by

a

comma

(,).

For

more

information

about

SIP

response

codes,

please

click

on

to

link

to

http://voip-info.org/wiki/view/sip+response+codes

where

you

can

find

out

the

meaning

of

each

error

code.

Advanced – PSTN Environment

Adjustment

The

PSTN

Environment

Adjustment

options

will

help

you

to

adjust

the

on

hook

and

off

hook

voltage

detection

values

for

your

environment.

The

actual

levels

are

determined

by

your

environment

including

the

number

and

type

of

telephones

used.

The

default

values

provided

are

suitable

for

the

South

African

PSTN

network,

and

there

should

not

be

modified.

If,

however,

you

are

connecting

the

line

port

to

a

PABX,

and

you

experience

problems

with

placing

calls,

then

you

may

wish

to

modify

these

parameters.

Note:

ON

HOOK

means

hung

up.

To

take

your

phone

OFF

HOOK,

lift

the

receiver

then

press

Hook/Flash

until

you

hear

your

normal

PSTN

dial

tone,

not

your

VoIP

dial

tone.

Wait

several

seconds

and

then

press

Check

Level.

You

should

check

the

OFF

HOOK

value

for

each

telephone

you

have

connected

to

this

device.

Set

the

OFFHOOK

voltage

to

the

lowest

setting

registered

for

all

your

telephones,

e.g.

if

your

telephones

return

values

of

4,

5

and

7

then

you

should

set

your

OFFHOOK

voltage

to

4.

Note:

The

detected

values

will

not

automatically

be

set

by

the

Check

Level

function;

you

must

enter

the

lowest

level

detected

after

testing

all

your

telephones.

1

03

Chapter

4:

Configuration

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Billion 800VGT Router

Phone

Port

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

is

the

registration

ID

of

the

user

as

recorded

in

the

VoIP

SIP

registrar.

Authentication

Username:

Same

as

Phone

Number.

Authentication

Password:

This

is

the

password

used

for

authentication

with

the

VoIP

SIP

registrar.

Confirm

Password:

Re-enter

the

password

for

confirmation.

Display

Name:

This

is

what

will

be

shown

when

using

the

Caller

ID

function

.

1

04

Chapter

4:

Configuration

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Billion 800VGT Router

Codec Preference

A

codec

is

a

Coder-Decoder

and

is

used

for

data

signal

conversion.

The

priority

position

sets

the

priority

of

each

codec;

Priority

1

is

the

top

priority.

G.729:

This

type

of

codec

encodes

and

decodes

the

voice

information

into

a

single

packet

which

reduces

the

bandwidth

consumption.

8kbps

of

bandwidth

is

needed.

G.711

μ

-LAW:

This

codec

uses

a

basic

non-compressed

encoder

and

decoder

technique.

μ

-LAW

uses

a

pulse

code

modulation

(PCM)

encoder

and

decoder

to

convert

voice

into

a

14-bit

linear

sample.

64kbps

of

bandwidth

is

needed.

G.711A-LAW:

This

codec

uses

a

basic

non-compressed

encoding

and

decoding

technique.

μ

-LAW

uses

a

pulse

code

modulation

(PCM)

encoder

and

decoder

to

convert

voice

into

a

13-bit

linear

sample.

64kbps

of

bandwidth

is

needed.

Non

-

used:

This

option

is

only

available

for

Priority

2

and

3.

It

should

be

selected

if

no

codec

is

to

be

used

in

these

priority

settings

.

Note:

In

the

example

screen

shown

above,

the

codec

priority

is

assigned

in

the

order

as

G.729

>

G.711

μ

-LAW

>

G.711A-LAW

.

Speed Dial

The

Speed

Dial

function

is

useful

for

storing

frequently

used

telephone

numbers.

You

can

press

a

number

from

0

to

9

and

the

hash

sign

(#)

on

the

phone

keypad

to

call

a

speed

dial

number.

For

example,

to

phone

a

speed

dial

number

listed

under

9,

press

keypad

9

then

#

.

Your

router

will

automatically

dial

the

number

listed

in

entry

9.

For

examples:

If

your

friend

Tim

gives

you

a

SIP

URL

as

sip:

[email protected]

then

you

can

fill

in

[email protected]

.

as

number

1

speed

dial.

If

your

friend

Felix

gives

you

a

SIP

URL

as

sip:

[email protected]

then

you

can

fill

in

[email protected]

.

as

number

2

speed

dial.

If

your

friend

Greg

gives

you

an

IP

address

"192.246.56.56"

only,

then

you

can

fill

in

“192.246.56.56”.

In

some

cases,

when

a

user

makes

use

of

DDNS,

you

will

have

to

fill

in

their

domain

name

as

well.

Volume Control

The

Volume

control

setting

helps

you

to

adjust

the

voice

level

of

the

telephone

to

the

most

comfortable

listening

level.

Press

“

-

“,

the

minus

sign,

to

reduce

either

the

microphone

and/or

the

speaker’s

volume

of

your

telephone.

Press

“

+

“,

the

plus

sign,

to

increase

either

the

microphone

and/or

the

speaker’s

volume

of

your

telephone.

1

05

Chapter

4:

Configuration

Downloaded from

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