TELES VoIP Box BRI Series, Reference Manual

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5 Additional VoIP parameters
VoIPBox DSL / VoIPBox BRI – version 21.0.009 Page 97
5.2 STUN server parameter
Enter here the STUN server’s IP address. A STUN server allows
NAT clients (i.e. computers behind a firewall) to setup phone
calls to a VOIP provider hosted outside of the local network.
When this parameter is active, the gateway looks for a (NAT)
firewall in the network and figures out how to bypass it. All
ports for signaling, RTP and RTCP are checked starting from
VoipSipLine=<string>
SIP only. Use this parameter to include a freely defined line in a SIP INVITE. This pa-
rameter can be used several times.
VoipSipStatusIncompatibleCodec=<reject cause>
An INVITE request is rejected if the codec that is sent with this INVITE does not match
the codecs that have been defined in the VoIP profile. The INVITE is rejected with the
reject cause given in this parameter, as per RFC 3261 (default 415).
VoipSipUserPhone=<mode>
SIP only. Set this parameter to Yes to add ;user=phone in the request header and the
To: line. Default is No.
VoipUseMaxPTime=<mode>
SIP only. Enter yes to set the field mptime (max packet time) with the values set in
VoipTxm (ptime). Default no.
The parameter VoipUseMaxPTime is used when VoipUseMPTime is 0, 1 or 2.
VoipUseMPTime=<int>
This parameter is used to configure packet time signaling in SDP:
0 = set attribute ptime with each individual codec description (default).
1 = set attribute ptime once as the first attribute after the m- line (media type).
2 = set attribute mptime (multiple ptime) once as the first attribute with the list of the
codecs’ corresponding ptimes.
3 = remove attribute ptime or mptime in SDP signaling.
The parameter VoipUseMaxPTime is used when VoipUseMPTime is 0, 1 or 2.
VoipUserAgent=<mode>/<string>
When Yes is set (default), the system type and software version is used in the SIP user
agent header. Enter no to deactivate this setting, or enter a string to change it.
VoipUtuFormat=<int>
Enter 1 to transfer ISDN user to user information to SIP and vice versa in the following
special format: A separate contents block is created (using boundaries) or evaluated
that contains the ISDN user to user info. Default is 0, which transports the ISDN user
to user information in a SIP header field "User-to-User:", according to the recommen-
dation in http://tools.ietf.org/html/draft-johnston-sipping-cc-uui-09.
Table 5.3 Customized parameters: SIP signaling (page 4 of 4)
SIP Signaling Parameters