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Status

The Status page displays ADSL status information

Parameter Description

Status

Line Status:

Shows the current status of the ADSL

line.

Data Rate:

Downstream:

Actual and maximum downstream

data rate.

Upstream:

Actual and maximum upstream data

rate.

Operation Data/Defect Indication:

Noise Margin Upstream:

Minimum noise margin

upstream

Downstream:

Minimum noise margin downstream

Output Power:

Maximum f luctuation in the output

power

Attenuation Upstream:

Maximum reduction in the

strength of the upstream signal

Attenuation Downstream:

Maximum reduction in the

strength of the downstream signal

Fast Path FEC:

There are two latency paths that

may be used: fast and Correction interleaved. For

either path a forward error correction (FEC) scheme

is employed to ensure higher data integrity. For

maximum noise immunity, an interleaver may be used

to supplement FEC. Interleaved Path An interleaver

is basically a buffer used to introduce a delay, FEC

Correction allowing for additional error correction

techniques to handle noise. Interleaving slows the

data flow and may not be optimal for real-time signals

such as video transmission.

Fast Path CRC indicates the number of Fast Path

Cyclic Redundancy Check Error errors. Interleaved

Path indicates the number of Interleaved Path Cyclic

Redundancy Error Check errors.

Loss of Signal Momentary signal discontinuities.

Defect Loss of Frame Failures due to loss of frames.

Loss of Power Defect:

Failures due to loss of power.

Fast Path HEC Error:

Fast Path Header Error

Concealment errors.

Interleaved Path HEC Error:

Interleaved Path Header

Error Concealment errors.

Statistics:

(Superframes represent the highest

level of data presentation which is used to provide

superframe synchronization, identifying the start of

a superframe. Some of the remaining frames are also

used for special functions).

Received Cells:

Number of interleaved superframes

received Interleaved.

Transmitted Cells:

Number of interleaved super

frames transmitted Superframes Interleaved.

Received Number of fast super frames received.

Superframes Fast

Transmitted Number of fast super frames transmitted.

Superframes Fast

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VoIP

Port Setting

Configure the port settings on this page, and click ‘SAVE SETTINGS’ to save the parameters. VoIP providers

operate SIP proxies that allow you to register your router on their system so that your can call friends, family

and business associates. Your BoB

TM

- 4 port integrated wireless router comes pre-configured for the iiNet VoIP

service. iiNet and Belkin will only provide support for use with the iiNet VoIP service.

See below for a description of the parameters.

Parameter Description

Phone 1/2 Enable:

Enable/disable phone 1 and/or 2.

Phone Number:

Your phone number.

Display Name:

Your name, often the same as your phone number.

SIP Domain:

(From your VoIP provider).

Sip Server:

(From your VoIP provider).

Username:

(From your VoIP provider).

Password:

(From your VoIP provider).

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SIP Setting

Configure your SIP parameters on this page, and click ‘SAVE SETTINGS’ to apply them.

SIP, the Session Initiation Protocol, is a signalling

protocol for Internet conferencing, telephony,

presence, events notification and instant messaging.

The call waiting feature allows the user to take an

incoming call, even though the user is already on the

phone. The user upon hearing the new call can put

the original caller on hold and speak to the new caller.

When the user has finished talking to the new caller,

he can resume his conversation with the original

caller.

According to the SIP RFC, a proxy server is ‘An

intermediary entity that acts as both a server and a

client for the purpose of making requests on behalf of

other clients. A proxy server primarily plays the role of

routing, which means its job is to ensure that request

is sent to another entity ‘closer’ to the targeted user.’

The proxy server therefore, is an intermediate device

that receives SIP requests from a client and then

forwards the requests on the client’s behalf. Proxy

servers receive SIP messages and forward them to

the next SIP server in the network. A series of proxy

and redirect servers receive requests from a client

and decide where to send these requests. Proxy

servers can provide functions such as authentication,

authorization, network access control, routing, reliable

request retransmission, and security.

From the SIP RFC, ‘A registrar is a server that accepts

REGISTER requests and places the information it

receives in those requests into the location service

for the domain it handles.’

See below for a description of the parameters.

Parameter Description

SIP Listen Port: It is strongly recommended that you

to leave the SIP port unchanged (Default: 5060).

Proxy Setting set the proxy settings.

Proxy IP: IP address of your proxy server. (From your

VoIP provider)

Proxy Port: Port number of the proxy server. (From

your VoIP provider)

Registrar Setting set the registrar settings.

Registrar IP: IP address of SIP registrar.

Registrar Port: Port number of SIP registrar.

•

•

•

•

•

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VoIP Advanced Setting

Configure the VoIP advanced settings on this page, and click ‘OK.’

SIP is a peer-to-peer protocol. The peers in a session

are called User Agents (UAs). A user agent can

function in one of the following roles:

User agent client (UAC) - A client application that

initiates the SIP request.

User agent server (UAS) - A server application that

contacts the user when a SIP request is received

and that returns a response on behalf of the user.

Typically, an SIP end point is capable of functioning as

both a UAC and a UAS, but functions only as one or

the other per transaction

Phone standards vary internationally and from

provider to provider, so it is important that the router

is configured correctly for your provider.

Codec are used to convert an analogue voice signal

to digitally encoded version. Codec vary in the sound

quality, the bandwidth required, the computational

requirements, etc. You can specify which audio

coding process you would like to use. There are four

voice codec supported by the router, you may try

different settings to determine the best audio quality

you obtain from the combination of your network

connection and your used audio device (head set or

hand set). The default codec sequence is listed below.

You can use the Up and Down buttons to change

priority.

G.711 A law

G.711 U law

1.

2.

1.

2.

See the below for a description of the parameters

Parameter Description

Support Call Waiting:

Enables or disables support for

call waiting (Default: Disabled).

Support User-Agent Header:

Enables or disables

user-agent header support. Enabling this feature

includes user agent information in the packet, e.g.,

the caller’s ID may be displayed. (Default: Disabled).

Telephony Hook F lash Timer:

The hook f lash timer is

the length of time before the hook f lash indicates a

time-out (or call disconnect).

(Default: 50 ~ 250 milliseconds)

Telephony Tone Country Setting:

Select the country

Voice Codec Configuration: Set the voice codecs.

Available Codecs:

List of available codecs.

Selected Codecs:

List of selected codecs, move

the preferred codec to the top of the list with up and

down buttons to the right. The codec at the top of

the list will be used when it can.

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Port Advanced Setting

Configure advanced VoIP settings on this page then click ‘SAVE SETTINGS’.

Volume Gain Control

Use this option to adjust the volume of calls made

through VoIP.

VAD

Voice Activation Detection. VAD is designed to

conserve bandwidth by halting transmission of voice

packets until it has detected a noise either by voice

or outside noise. The downside to this is it may miss

some packets due to a slight delay in the transmission

of packets. Disable this if you are experiencing issues

with phone system menus, faxing over IP, etc.

Caller ID Mode

Use DTMF Caller ID Mode. Enabling this option enabled

the Dual Tone, Multi-Frequency (touch tone) mode for

Caller ID

Inter Digit Delay

This is the delay time before processing the dialled

digits This will delay the VoIP unit dial the telephone

number after the digits have been entered.

T.38 Mode

38 is the standard for sending faxes over IP networks.

Enable this option for Faxing over IP.

Dial Tone(Hz)

Adjust the pitch of the VoIP dial tone.

Dialling Plans

Configure the VoIP dialling plans on this page, and

click ‘SAVE SETTINGS’.

Set the Phone Number and Connection Type on this page.

Chapter 5 :

Advanced Setup

BoB

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Advanced Setup Method