DBL Technology GoIP-1 User Manual page 30

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3. Call OUT Auth. Mode
4. Built-in SIP Proxy

Password
5. NAT Keep-Alive
6. DTMF Signaling
 Outband DTMF Type
 RTP Payload Type
7. Signaling QoS
8. Signal Encryption
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media (before the call is answered). This allows to the caller to hear the ringback
from the GSM network. This is done in order to avoid a long silent period before
a ringback tone is returned from the GSM network.
3. SIP 183 - The device sends back a SIP 183 Session In Progress message to the
calling SIP device. The calling SIP device then goes into early media mode to
receive audio packets. Since it may take a few to over 10 seconds for a ringback
tone is returned from the GSM network, the caller may hear a long silent period
This setting defines how incoming VoIP calls are authenticated when the device is
configured for using SIP registration(s). This setting applies to Single Server Mode,
Config. By Line and Config. by Group modes. This prevents unauthorized calls to be
dialed out via the GSM Channel(s). The following authentication methods are
available:
1. None - No authentication is used for calls received. This could be a simple
arrangement if calls are routed from a SIP Server in the same local network.
2. IP – only calls received from the registered SIP Server(s) are accepted.
3. Password – A SIP 401 message is sent to the SIP server for password
authentication of the corresponding SIP account when a call is received.
4. IP and Password – Both authentication methods are used.
A simple SIP Proxy is embedded in the device. Choose "Enable" to activate this SIP
proxy to accept any SIP registrations with the correct password which is specified in the
parameter "Password". There is no need to create a SIP account in this server. Users
will have to manage the SIP numbers used on their own. This facilitates the setup of a
simple SIP network for customers who do not have their own SIP servers.
This sets the password for SIP registration to the built-in SIP server.
When enabled, NAT Keep Alive sends a NULL packet to the router regularly in order to
keep the network ports used open.
This setting specifies the DTMF dialing method.
1. Inband – DTMF tones are generated in the form of audio stream.
2. Outband – DTMF digits are sent in the form of digital commands (RFC2833 / SIP
INFO). DTMF tones are actually generated by the terminating party.
This parameter is for outband DTMF dialing. Select the proper format (RFC2833 or SIP
INFO) as required by your SIP network.
This parameter specifies the payload type in RFC 2833 commands.
This specifies the QoS method used for SIP signaling. Both IP TOS and DiffServe format
are supported. Select the proper setting that is compatible with your network
environment.
Signaling encryption is employed to offer a more secure environment for SIP
communications. The following encryption methods are supported. Consult your
network/VoIP administrator for more the proper selection if required.
1. RC4
2. Fast
3. VOS
4. AVS
5. N2C
6. ECM
7. ET263
8. XOR
GoIP User Manual
Technical Support: support@dbltek.com
IP
Disabled
Enabled
Outband
2833
101
None
None
30