Belkin F1PI24EGau Router Manual PDF (Setup & Configuration Guide)

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Advanced Setup Method

section

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, a 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 VoIP router is configured correctly for your

provider.

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 VoIP 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.729

G.723.1

G.711 U law

G.711 A law

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 Flash Timer:

The hook flash timer is the length of

time before the hook flash indicates a time-out (or call disconnect).

(Default: 50 ~ 250 milliseconds.)

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).

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.

VoIP Advanced Setting

Configure the VoIP advanced settings on this page, and click “SAVE

SETTINGS”.

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Advanced Setup Method

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Jitter Buffer Mode

Jitter Buffer helps eliminate jitter caused by transmission delays on

a VoIP call. As the jitter buffer receives the voice packets it adds

a small amount of delay to the packets so it appears they were all

received without delays.

NONE – Jitter Buffer is disabled

FIXED – Jitter Buffer Mode is fixed

ADAPTIVE – Jitter Buffer Mode will automatically adapt to the current

call.

SEQUENTIAL – Jitter Buffer Mode set to Sequential

Jitter Buffer Delay

Specifies the delay in milliseconds for the Jitter Buffer. Default/

recommended is 40ms.

Echo Canceller Delay

Echo cancellation is the process in removing echo from voice

communication over the VoIP. It improves the quality of the call and

conserves bandwidth.

Default/recommended setting = 32 milliseconds

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.

Default/recommended = Enabled

CNG

Comfort Noise Generation. As VoIP is digital, there is no background

interference like there is on the standard analogue PSTN (Public

Switched Telephone Network). This option will generate slight

noises in the background to make the digital call sound more like an

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.

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:

OFF – Standard volume level 0dB.

FIXED – Set the volume to amplify or attenuate at a fixed dB.

ADAPTIVE – The volume will automatically amplify or attenuate

according to the current call.

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Advanced Setup Method

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Dialing Plans

Configure the VoIP dialing plans on this page, and click “SAVE

SETTINGS”.

Set the Phone Number and Connection Type on this page.

VoIP Status and Call Logs

View the VoIP status for both FXS ports on this page. Click “Refresh”

to update this page.

analogue call. Disable this if you are experiencing issues with phone

system menus, Faxing over IP etc.

Default/recommended = Enabled

PLC

Packet Loss Compensation. PLC is used only when utilising the

G.711 codec, the algorithm is designed to compensate for loss

packets. Re-transmitting the lost packets is obviously not a viable

option with a digital VoIP telephone call.

Default/recommended = Enabled

Caller ID Mode

Use DTMF Caller ID Mode. Enabling this option enables the Dual

Tone, Multi-Frequency (touch tone) mode for Caller ID.

Default/recommended = Disabled

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.

Default/recommended = 4 Seconds

Additional Ringing Mode

Enabling this option will force the VoIP telephone to ring when an

incoming call is made through via the PSTN number. You will need to

have a filtered telephone cable connected to the PSTN Failover.

Default/recommended = Enabled

T.38 Mode

T.38 is the standard for sending faxes over IP networks. Enable this

option for Faxing over IP.

Default/recommended = Enabled

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Advanced Setup Method

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UPnP

The Universal Plug and Play architecture offers pervasive peer-to-peer

network connectivity of PCs of all form factors, intelligent appliances,

and wireless devices. UPnP enables seamless proximity network in

addition to control and data transfer among networked devices in the

home, office and everywhere in between.

Enable or disable UPnP features: Enable or disable the UPnP function.

QoS

With converging voice and data, it is imperative to establish Quality

of Service (QoS) parameters to appropriately allocate bandwidth. QoS

will only monitor and limit upstream traffic.

QoS Settings

To ensure optimum voice quality, your VoIP Router should prioritize

voice over data packets. Therefore, we recommend enabling the QoS

feature.

This page displays the Port Type, SIP URL and Registration status of

the VoIP router.

See the table below for a description of the parameters.

Parameter

Description

Port Type

Displays the port type, i.e., FXS.

SIP URL

Shows the SIP URL.

Registration

Indicates whether the user has successfully

registered or not.

VoIP Call Logs

View the call log for both FXS ports on this page. Click “Refresh” to

update the page.

See the table below for a description of the parameters.

Parameter

Description

Port Type

Displays the port type, i.e. FXS.

Received Call

Number of received calls.

Dialed Call

Number of calls made.

Rejected Call

Number of rejected calls.

Forwarded Call

Number of forwarded calls.

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Advanced Setup Method

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Traffic Mapping

Up to 16 rules can be defined to classify traffic into Diffserv

forwarding groups and outgoing VCs.

Click on “Add Traffic Class” or click on “Edit” and a mapping already

in the list to bring up the following screen and enter a setting which is

to be mapped to a QOS group.

Parameter Description

Enable or disable. QoS module function:

Enables or disables QoS

Diffserv Forwarding Groups:

You can set the minimum amount

of bandwidth you want allocated for certain QOS groups in a

Percentage. The different groups allow you to manage your different

types of connections more efficiently.

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