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Page 36 / 52 Scroll up to view Page 31 - 35
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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, 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.
1.
G.729
2.
G.723.1
3.
G.711 U law
4.
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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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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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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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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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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