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Configuring SIP settings

Configuring SIP settings

The FortiVoice Gateway supports SIP communications.

To configure SIP settings
  1. Go to System > Advanced Setting > SIP.
  2. Configure the following fields:

    GUI field

    Description

    SIP Transport

    Enable the ports, as required.

    SIP communication commonly uses TCP or UDP port 5060 and/or 5061. Port 5060 is used for nonencrypted SIP signaling sessions and port 5061 is typically used for SIP sessions encrypted with Transport Layer Security (TLS).

    RTP Setting

    Port

    Enter the start and end RTP ports that the FortiVoice Gateway will use for phone call sessions.

    Make sure that there is a reasonable port range so that you have enough ports for all open calls. The default port range is 10000 to 30000.

    RTP timeout

    Enter the amount of time in seconds during an active call that the extension will wait for RTP packets before hanging up the call. 0 means no time limit. The default is 60.

    RTP hold timeout

    Enter the amount of time in seconds that the extension will wait on hold for RTP packets before hanging up the call. 0 means no time limit. The default is 300.

  3. Click Apply.

Configuring SIP settings

Configuring SIP settings

The FortiVoice Gateway supports SIP communications.

To configure SIP settings
  1. Go to System > Advanced Setting > SIP.
  2. Configure the following fields:

    GUI field

    Description

    SIP Transport

    Enable the ports, as required.

    SIP communication commonly uses TCP or UDP port 5060 and/or 5061. Port 5060 is used for nonencrypted SIP signaling sessions and port 5061 is typically used for SIP sessions encrypted with Transport Layer Security (TLS).

    RTP Setting

    Port

    Enter the start and end RTP ports that the FortiVoice Gateway will use for phone call sessions.

    Make sure that there is a reasonable port range so that you have enough ports for all open calls. The default port range is 10000 to 30000.

    RTP timeout

    Enter the amount of time in seconds during an active call that the extension will wait for RTP packets before hanging up the call. 0 means no time limit. The default is 60.

    RTP hold timeout

    Enter the amount of time in seconds that the extension will wait on hold for RTP packets before hanging up the call. 0 means no time limit. The default is 300.

  3. Click Apply.