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Loss of Frame Defect Shows the current status of the ADSL line.
Actual and maximum upstream data rate.
Actual and maximum downstream data rate.
Minimum noise margin upstream.
Minimum noise margin downstream.
Maximum fluctuation in the output power.
Maximum reduction in the strength of the upstream signal.
Maximum reduction in the strength of the downstream signal.
There are two latency paths that may be used: fast and 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.
An interleaver is basically a buffer used to introduce a delay, 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.
Indicates the number of Fast Path Cyclic Redundancy Check errors.
Indicates the number of Interleaved Path Cyclic Redundancy Check errors.
Momentary signal discontinuities.
Failures due to loss of frames.
Parameter Description
Loss of Power Defect
Fast Path HEC Error
Interleaved Path HEC Error
Statistics
Received
Superframes
Interleaved
Transmitted
Superframes
Interleaved
Received Superframes Fast
Transmitted Superframes Fast Failures due to loss of power.
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Fast Path Header Error Concealment errors.
Interleaved Path Header Error Concealment errors.
(Superframes represent the highest level of data presentation. Each superframe contains regular
ADSL frames, one of which is used to provide superframe synchronization, identifying the start of
a superframe. Some of the remaining frames are also used for special functions.)
Number of interleaved superframes received.
Number of interleaved superframes transmitted.
Number of fast superframes received. Number of fast superframes transmitted.
Telephony
Port Setting
Configure the port settings on this page, and click “OK” to save the parameters. Telephony
providers operate SIP proxies that allow you to register your Belgacom b-box on their
system so that your can call friends, family and business associates. There are many
Telephony service providers available. It is up to you to decide which service provider is
best for your needs. Once you have decided on a provider, you will need to get the
following information:
Username
Password
SIP Domain
Realm
SIP Proxy IP
SIP Proxy Port
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See the table 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 Telephony provider, e.g., “sipcenter.com” or an IP address.)
Realm (From your Telephony provider.)
Username (From your Telephony provider.)
Password (From your Telephony provider.)
SIP Setting
Configure your SIP parameters on this page, and click “SAVE” to apply them.
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SIP, the Session Initiation Protocol, is a signaling 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 hasn’t
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 the table 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).
Support Call Enables or disables support for call waiting. Waiting (Default: Disabled)
Proxy Setting Set the proxy settings.
±
Proxy IP: IP address of your proxy server. (From your Telephony provider.)
±
Proxy Port: Port number of the proxy server. (From your Telephony provider.)
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Registrar Setting Set the registrar settings.
±
Registrar IP: IP address of SIP registrar.
±
Registrar Port: Port number of SIP registrar.
Telephony Advanced Setting
Configure the Telephony 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:
1.
User agent client (UAC) - A client application that initiates the SIP request.
2.
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, so it is important that the Belgacom b-box is configured
for the correct country.
Codecs are used to convert an analog voice signal to digitally encoded version. Codecs 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 codecs supported by the
Belgacom b-box, 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.
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