2N Clip User Manual page 37

2N® Clip User Manual
PRACK Allowed – enable the PRACK method (reliable confirmation of SIP messages with
codes 101–199).
REFER Allowed – enable the REFER method for call redirection.
Send Keep Alive Packets – define whether or not the device shall periodically inquire
about the called station status via SIP OPTIONS requests during calls (used for detection of
the station failure during calls).
Allow IP Address Filter – enable the blocking of SIP packet receiving from addresses other
than the SIP Proxy and SIP Registrar. The primary purpose of the function is to enhance
communication security and eliminate unauthorized phone calls.
Accept Encrypted Calls Only (SRTP) – set that the SRTP encrypted calls shall only 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.
Encrypted Outgoing Calls (SRTP) – set that outgoing calls shall be SRTP encrypted on this
account. At the same time, TLS is recommended as the SIP transport protocol for higher
security.
Use MKI in SRTP Packets – enable the use of the MKI (Master Key Identifier), which is
required by the counterparty for primary key identification if multiple keys are rotated in
SRTP packets.
Do Not Play Incoming Early Media – disable playing of the incoming video stream before
the call is picked up (early media), which is sent by some PBXs or other devices. A standard
local ringtone is played instead.
QoS DSCP Value – set the priority of SIP packets in the network. The set value is sent in the
TOS (Type of Service) field in the IP packet header. Enter the value as a decimal number. A
change of this parameter will not be applied until the device is restarted.
External IP Address – set the public IP address or router name to which the device is
connected. If the device IP address is public, leave this parameter empty.
Starting RTP Port – set the initial local RTP port in the range of 64 ports used for audio and
video transmission. The default value is 4900 (i.e. the range is 4900–4963). The parameter is
only set for SIP 1 but applies to both the SIP accounts.
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 will be terminated by the device. Enter
0 to disable this parameter. The parameter is only set for account 1 but applies to both the
SIP accounts.
Broadsoft PBX Compatibility – set the Broadsoft PBX compatibility mode. Having received
re-invite from a PBX in this mode, the intercom replies by repeating the last sent SDP with
currently used codecs instead of sending a complete offer.
SRV Record Rotation – allow SRV record rotation for SIP Proxy and Registrar. This is an
alternative method of transition to backup servers in the event of main server failure or
unavailability.
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