Matrix PRASAR UCS System Manual 173
• Local Menu
Transport Mode and SRTP
• Select the protocol to be used to transport the SIP messages. You can select the Transport Mode as TCP
or TLS.
• If you select TCP, make sure the SIP Over TCP check box is selected in VoIP Parameters.
• If you select TLS, make sure the SIP Over TLS check box is selected in VoIP Parameters.
• For secure conversations over SIP, select the Enable SRTP? check box. The SIP messages will be
transported over SRTP only.
If you select this check box, make sure you have selected SRTP Mode as Forced or Optional in the
General Parameters under SIP Extension Settings.
RTP Port
• Define RTP Port:
• RTP Listening Port: This is the port on which the SPARSH VP310 listens for RTP messages over
UDP. This port is also used as the source port for sending RTP packets. This port is also used as the
source port for sending RTP packets to the remote peer. The valid range for this port is 1025-65278.
Default: 8000.
Quality of Service
• Set the SIP Quality of Service (QoS) for SIP signaling as:
• SIP DiffServe/ToS. Valid range is 00 to 63. Default: 26.
OR
• RTP DiffServe/ToS. Valid range is 00 to 63. Default: 46.
NAT Keep Alive
• If the SPARSH VP310 is connected behind a NAT router, configure NAT Keep Alive.
• Select the Enable NAT Keep Alive check box to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
• Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
The time period you define should be less than the binding timer of the router.
Timers
• Set the following Timers to the desired value, where required:
• SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after sending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user
disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
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