Operation Manual

Chapter 14 VoIP
LTE7410 User’s Guide
108
SIP Service
Domain
Enter the SIP service domain name. In the full SIP URI, this is the part after the @
symbol. You can use up to 127 printable ASCII Extended set characters.
Bound Interface Name
Bound Interface
Name
If you select AnyWAN, the LTE Device automatically activates the VoIP service when any
WAN connection is up.
If you select MultiWAN, you also need to select the pre-configured WAN connections. The
VoIP service is activated only when one of the selected WAN connections is up.
RFC Support
PRACK (RFC
3262)
RFC 3262 defines a mechanism to provide reliable transmission of SIP provisional
response messages, which convey information on the processing progress of the request.
This uses the option tag 100rel and the Provisional Response ACKnowledgement (PRACK)
method.
Select Supported or Required to have the LTE Device include a SIP Require/Supported
header field with the option tag 100rel in all INVITE requests. When the LTE Device
receives a SIP response message indicating that the phone it called is ringing, the LTE
Device sends a PRACK message to have both sides confirm the message is received.
If you select Supported, the peer device supports the option tag 100rel to send
provisional responses reliably.
If you select Required, the peer device requires the option tag 100rel to send provisional
responses reliably.
Select Disabled to turn off this function.
DNS SRV Enabled
(RFC 3263)
Select this to have the LTE Device query your ISP’s DNS server for a list of any available
SIP servers that it maintains. This is useful if your static SIP server experiences
difficulties, making it hard for your IP phone users to make SIP calls.
Session Timer
(RFC 4028)
Select this to have the LTE Device support RFC 4028.
This makes sure that SIP sessions do not hang and the SIP line can always be available for
use.
VoIP IOP Flags - Select VoIP inter-operability settings.
Replace dial digit '#' to '%23' in SIP messages.
Remove ':5060' and 'transport=udp' from request-uri in SIP messages.
Remove the 'Route' header in SIP messages.
Don't send re-Invite to the remote party when there are multiple codecs answered in the
Session Description Protocol (SDP).
Remove the 'Authentication' header in SIP ACK messages.
RTP Port Range
Start Port
End Port
Enter the listening port number(s) for RTP traffic, if your VoIP service provider gave you
this information. Otherwise, keep the default values.
To enter one port number, enter the port number in the Start Port and End Port fields.
To enter a range of ports,
• enter the port number at the beginning of the range in the Start Port field.
• enter the port number at the end of the range in the End Port field.
Table 41 VoIP > SIP > SIP Service Provider > Edit (continued)
LABEL DESCRIPTION