
Dialogic® Diva® SIPcontrol™ Configuration
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Security
You can configure the following parameters in the Security section when you define or modify a SIP Peer:
Session Timer
You can configure the following parameters in the Session Timer section when you define or modify a SIP Peer:
Reliable
provisional
response:
SIP defines two types of responses, provisional and final. Provisional responses provide information
on the progress of the request processing and final responses transmit the result of the request
processing.
This parameter specifies whether reliable provisional responses (RFC 3262) should be used. The
following values are available:
• Disabled: Reliable provisional response is not used.
• Optional: Reliable provisional response can be used.
• Required: Reliable provisional response is mandatory.
Signaling accept level: This parameter defines how call information should be accepted. To accept encrypted calls,
you need to activate TLS as listen port in the Network Interfaces
configuration.
• Accept unencrypted calls only: Only signaling sent with TCP or UDP is accepted. Any
encrypted signaling is rejected.
• Accept encrypted and unencrypted calls: All calls are accepted, regardless of the
encryption mode.
• Accept encrypted calls only: Only signaling with TLS is accepted; unencrypted signaling
is rejected.
• Accept encrypted call with SIPS URI only: Only signaling encrypted with the URI
scheme secure SIP is accepted. Calls sent with TLS encryption are rejected.
Media security level: The Secure Real-time Transport Protocol (SRTP) authenticates packets and encrypts data and
thus adds security to the voice stream. SRTP should be used together with TLS.
• No SRTP: The voice stream is not secured with SRTP.
• Offer and accept SRTP: The voice stream is secured with SRTP, if possible.
• Require SRTP for encrypted calls: Calls via TLS need to use SRTP, otherwise they are
rejected.
• Require STP for all calls: All calls are established with SRTP only, regardless of the
signaling protocol.
Note: If you select Require SRTP for encrypted calls, calls without SRTP are still allowed
via UDP or TCP, unless Signaling accept level does not allow calls via UDP or TCP.
Use session timer: Activates session monitoring via SIP session timers using the time-out values given here. Refer
to RFC 4028 for details.
Interval: If Use session timer is enabled, you can set a time-out in seconds until a call is considered to
be aborted. Refreshes are normally performed after the first half of the interval has elapsed. The
minimum value is 90 seconds. The default value is 600 seconds.
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