Matrix NAVAN CNX200, System Manual

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412 Matrix NAVAN CNX200 System Manual
• If you select TCP, make sure that Enable SIP Over TCP check box is selected in VoIP Port page
located under PBX Settings tab. For details, see “VoIP Port” .
• If you select TLS, make sure that Enable SIP Over TLS check box is selected in VoIP Port page
located under PBX Settings tab. For details, see “VoIP Port” .
• Enable SRTP?: For secured conversations, select the Enable SRTP? check box. It offers confidentiality,
message authentication, and replay protection for the RTP traffic which prevents eavesdropping and
improves the voice quality.
RTP Port
• Define RTP Port :
• RTP Listening Port: This is the port on which the IP extension listens for 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 (QoS)
• Set the SIP Quality of Service (QoS) for SIP signaling as:
• SIP DiffServe/ToS . Valid range is 00 to 63. Default: 26.
• RTP DiffServe/ToS . Valid range is 00 to 63. Default: 46.
NAT Keep Alive
• If the Soft-phone Client device 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 9999 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 Soft-phone Client 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 Soft-phone Client terminates the call process and gives
error tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec) : This is the time in seconds that the Soft-phone Client waits for final
response after receiving the provisional response from the called party. This timer starts on the receipt
of the provisional response from the called party and stops on receipt of the final response from the
called party or when the user disconnects the call. On expiry of the timer, the Soft-phone Client