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Page 46 / 71 Scroll up to view Page 41 - 45
Page 43
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
Chapter 5 :
Advanced Setup
BoB
TM
Advanced Setup Method
Page 47 / 71
Page 44
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).
Chapter 5 :
Advanced Setup
BoB
TM
Advanced Setup Method
Page 48 / 71
Page 45
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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Chapter 5 :
Advanced Setup
BoB
TM
Advanced Setup Method
Page 49 / 71
Page 46
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.
Chapter 5 :
Advanced Setup
BoB
TM
Advanced Setup Method
Page 50 / 71
Page 47
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
TM
Advanced Setup Method

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