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FortiVoice Gateway Administration Guide

Configuring SIP profiles

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Configuring SIP profiles

Configure the SIP related settings and codecs and apply them to SIP trunks.

Note

Communicate with your VoIP service provider because the profile settings are subject to the capabilities of the VoIP service provider. For example, if some of your features and codecs are not supported by your VoIP service provider, they will not work even if they are enabled or selected in the SIP profile.

You can edit the default SIP profiles but you cannot delete them.

To configure a SIP profile

  1. Go to Gateway > SIP > Profile.
  2. Click New.
  3. Select Trunk.
  4. Configure the following fields:

    GUI field

    Description

    Name

    Enter a name for this profile.

    DTMF

    Select the dual-tone multi-frequency (DTMF) method used by the VoIP provider. Options are RFC2833, Inband, Info, Shortinfo, and Auto.

    Auto means the VoIP provider’s server and the FortiVoice Gateway will negotiate to select a DTMF method. You could also select a specific DTMF method if required.

    Keep alive

    Enter the time interval in seconds for the FortiVoice Gateway to talk to the SIP server of your service provider to keep the connectivity and check its capability.
    A value of 0 (zero) means no checking by the FortiVoice Gateway.

    NAT

    Select if the VoIP service provider supports SIP NAT translation.

    T.38

    Select if the VoIP service provider supports fax over VoIP network.

    Registration interval

    To keep the extensions' registration status with the FortiVoice unit, keep the default value of extension registration time interval or enter the value in seconds as required by the FortiVoice unit. The default is 1800. The range is 10 - 28800.

    Subscription interval

    To keep the extensions’ subscription status with the FortiVoice unit, keep the default value of extension subscription time interval or enter the value in minutes as required by the FortiVoice unit. . The default is 60. The range is 10 - 1440.

    Transport

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

    Enable the protocols as required.

    Secure RTP: Select to provide encryption, message authentication and integrity, and replay protection to the FortiVoice Gateway Real-time Transport Protocol data.

    Codec

    Select the audio and video codecs supported by the VoIP service provider. Among the selected ones, choose the preferred one for the VoIP provider. The preferred codec is usually the most used one in your area and provides the best quality of communication.

    If your preferred codec is different from that of your VoIP service provider, the service provider’s codec will be used as long as it is one of your supported codecs.

  5. Click Create.

Configuring SIP profiles

Configure the SIP related settings and codecs and apply them to SIP trunks.

Note

Communicate with your VoIP service provider because the profile settings are subject to the capabilities of the VoIP service provider. For example, if some of your features and codecs are not supported by your VoIP service provider, they will not work even if they are enabled or selected in the SIP profile.

You can edit the default SIP profiles but you cannot delete them.

To configure a SIP profile

  1. Go to Gateway > SIP > Profile.
  2. Click New.
  3. Select Trunk.
  4. Configure the following fields:

    GUI field

    Description

    Name

    Enter a name for this profile.

    DTMF

    Select the dual-tone multi-frequency (DTMF) method used by the VoIP provider. Options are RFC2833, Inband, Info, Shortinfo, and Auto.

    Auto means the VoIP provider’s server and the FortiVoice Gateway will negotiate to select a DTMF method. You could also select a specific DTMF method if required.

    Keep alive

    Enter the time interval in seconds for the FortiVoice Gateway to talk to the SIP server of your service provider to keep the connectivity and check its capability.
    A value of 0 (zero) means no checking by the FortiVoice Gateway.

    NAT

    Select if the VoIP service provider supports SIP NAT translation.

    T.38

    Select if the VoIP service provider supports fax over VoIP network.

    Registration interval

    To keep the extensions' registration status with the FortiVoice unit, keep the default value of extension registration time interval or enter the value in seconds as required by the FortiVoice unit. The default is 1800. The range is 10 - 28800.

    Subscription interval

    To keep the extensions’ subscription status with the FortiVoice unit, keep the default value of extension subscription time interval or enter the value in minutes as required by the FortiVoice unit. . The default is 60. The range is 10 - 1440.

    Transport

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

    Enable the protocols as required.

    Secure RTP: Select to provide encryption, message authentication and integrity, and replay protection to the FortiVoice Gateway Real-time Transport Protocol data.

    Codec

    Select the audio and video codecs supported by the VoIP service provider. Among the selected ones, choose the preferred one for the VoIP provider. The preferred codec is usually the most used one in your area and provides the best quality of communication.

    If your preferred codec is different from that of your VoIP service provider, the service provider’s codec will be used as long as it is one of your supported codecs.

  5. Click Create.