TP-LINK TD-VG3631 User Manual
TD-VG3631
300Mbps Wireless N VoIP ADSL2+ Modem Router User Guide
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DSCP for SIP/RTP: DSCP(Differentiated Services Code Point) is the first 6 bits in the ToS
byte. DSCP marking allows users to assign specific application traffic to be executed in
priority by the next Router based on the DSCP value. Select DSCP for the SIP(
priority by the next Router based on the DSCP value. Select DSCP for the SIP(
Session
Initiation Protocol
) and RTP(Real-time Transport Protocol) respectively. If you are unsure,
please always keep the default value.
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Dtmf Relay setting: DTMF is Dual Tone Multi Frequency. Options available are SIPInfo,
RFC2833, and InBand. If you are unsure which one to choose, please always keep the
default value.
default value.
•
SIPInfo – If it is selected, the Router will capture the DTMF tone and transfer it into SIP
form. Then it will be sent to the remote end with SIP message.
form. Then it will be sent to the remote end with SIP message.
•
RFC2833 – If it is selected, the Router will capture the keypad number you pressed and
transfer it into digital form then send to the other side; the receiver will generate the tone
according to the digital form it receives. This function is very useful when the network
traffic congestion occurs and it still can remain the accuracy of DTMF tone.
transfer it into digital form then send to the other side; the receiver will generate the tone
according to the digital form it receives. This function is very useful when the network
traffic congestion occurs and it still can remain the accuracy of DTMF tone.
•
InBand – If it is selected, the Router will send the DTMF tone as audio directly when you
press the keypad on the phone.
press the keypad on the phone.
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Registration Expire Timeout(s): Expire time for the registration message sending.
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Registration Retry Interval(s): Set the time duration for your SIP Registrar server to keep
your registration record. Before the time expires, the Modem Router will send another register
request to SIP Registrar again. If you are unsure of it, please always keep the default value.
request to SIP Registrar again. If you are unsure of it, please always keep the default value.
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Enable T38 support: T38 specifies a protocol for transmitting a fax across IP network in real
time. It allows the transfer of fax documents in real-time between two standard Group 3
facsimile terminals over the Internet or other networks using IP protocols. It will only function
when both sites support this feature and are enabled.
facsimile terminals over the Internet or other networks using IP protocols. It will only function
when both sites support this feature and are enabled.
PSTN Setup
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Incoming PSTN Calls Routing: Choose a method for the Router to decide where to route
the incoming call on PSTN network.
•
Phone-Physical Phone – The Router will route the incoming call according to the
physical line selected in “Route Incoming PSTN Calls To” field. For example, if “Idle
Phone” is selected, the Router will first judge the status of Phone 1. If it is free, then the
incoming call will be routed to Phone 1. Otherwise, the Router will go on judging the
status of Phone 2.
physical line selected in “Route Incoming PSTN Calls To” field. For example, if “Idle
Phone” is selected, the Router will first judge the status of Phone 1. If it is free, then the
incoming call will be routed to Phone 1. Otherwise, the Router will go on judging the
status of Phone 2.
•
VoIP-VoIP call – The Router will route the incoming call according to the SIP number
and account set in “SIP URI of PSTN endpoint” field. For example, if “2654321@test1” is
input, the Router will directly route the incoming call to 2654321 through test1. The SIP
number and account can be a remote one, say a friend’s SIP number and account.
and account set in “SIP URI of PSTN endpoint” field. For example, if “2654321@test1” is
input, the Router will directly route the incoming call to 2654321 through test1. The SIP
number and account can be a remote one, say a friend’s SIP number and account.
Besides, VoIP-VoIP call allows second dialing when the SIP number but not account is
left blank, for example “@test1”. This function is applicable on the FXO ports that function
as a bridge between VoIP call and PSTN. The user can remotely use the VoIP line to
initial a call.
left blank, for example “@test1”. This function is applicable on the FXO ports that function
as a bridge between VoIP call and PSTN. The user can remotely use the VoIP line to
initial a call.
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PSTN polarity reversal support: This feature is disabled by default. If you are unsure of it,
please always keep the default.