User certificate – specify the user certificate and private key to verify the
intercom authorisation to communicate with the SIP server. There are three sets
of user certificates and private keys, refer to the Certificates subsection.
Local SIP port – set the local port to be used for SIP signalling. The parameter is
not applied until the intercom is restarted. The default value is 5060.
PRACK enabled enable the PRACK method for reliable confirmation of SIP –
messages with codes 101–199 .
REFER enabled – enable call forwarding via the REFER method.
Send KeepAlive packets – define whether the intercom shall, during a call, send
periodical SIP OPTIONS requests to inquire about the state of the called station
(to detect the station failure, e.g.).
IP address filter enabled – enable the blocking of SIP packet receiving from
addresses other than SIP Proxy and SIP Registrar. The primary purpose of the
function is to enhance communication security and eliminate unauthorised
phone calls.
– set that SRTP encrypted calls shall only Receive encrypted calls only (SRTP)
be received on this account. Unencrypted calls will be rejected. At the same
time, TLS is recommended as the SIP transport protocol for higher security.
– set that outgoing calls shall be SRTP Encrypted outgoing calls (SRTP)
encrypted on this account. At the same time, TLS is recommended as the SIP
transport protocol for higher security.
QoS DSCP value – set the SIP packet priority in the network. The set value is
sent in the TOS (Type of Service) field in the IP packet header. Value is entered
in decimal format. The parameter is not applied until the intercom is restarted.
Starting RTP port – set the starting local RTP port in the range of the length of
60 ports to be used for audio and video transmissions. The default value is 5000
(i.e. the used range is 5060–5059). The parameter is only set for account 1 but
applies to both the SIP accounts.
set the public IP address or name of the router to which External IP address –
your intercom is connected. If the intercom IP address is public, leave this field
blank.
RTP timeout – set the audio stream RTP packet receiving timeout during a call. If
this limit is exceeded (RTP packets are not delivered), the call is terminated by
the intercom. Set the parameter to 0 to disable this function. The parameter is
only set for account 1 but applies to both the SIP accounts.
Calls
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