AudioCodes Mediant 3000 User Manual page 305

SIP User's Manual
Parameter
Web: SIP TLS Local Port
EMS: TLS Local SIP Port
[TLSLocalSIPPort]
Web/EMS: Enable SIPS
[EnableSIPS]
Web/EMS: Enable TCP
Connection Reuse
[EnableTCPConnectionReu
se]
Web/EMS: TCP Timeout
[SIPTCPTimeout]
Web: SIP Destination Port
EMS: Destination Port
[SIPDestinationPort]
Web: Use user=phone in SIP
URL
EMS: Is User Phone
[IsUserPhone]
Web: Use user=phone in
From Header
EMS: Is User Phone In From
[IsUserPhoneInFrom]
Use Tel URI for Asserted
Identity
[UseTelURIForAssertedID]
Web: Tel to IP No Answer
Timeout
EMS: IP Alert Timeout
[IPAlertTimeout]
Version 5.8
Local TLS port for SIP messages.
The valid range is 1 to 65535. The default value is 5061.
Note: The value of must be different than the value of 'SIP TCP
Local Port' (TCPLocalSIPPort).
Enables secured SIP (SIPS URI) connections over multiple hops.
[0] Disable (default).
[1] Enable.
When 'SIP Transport Type' is set to TLS (SIPTransportType = 2) and
'Enable SIPS' is disabled, TLS is used for the next network hop only.
When 'SIP Transport Type' is set to TCP or TLS (SIPTransportType
= 2 or 1) and 'Enable SIPS' is enabled, TLS is used through the
entire connection (over multiple hops).
Note: If this parameter is enabled and 'SIP Transport Type' is set to
UDP (SIPTransportType = 0), the connection fails.
Enables the reuse of the same TCP connection for all calls to the
same destination.
[0] Disable = Use a separate TCP connection for each call.
[1] Enable = Use the same TCP connection for all calls (default).
Note: This parameter must be left at its default value (and not set to
0).
Defines the Timer B (INVITE transaction timeout timer) and Timer F
(non-INVITE transaction timeout timer), as defined in RFC 3261,
when the SIP Transport Type is TCP.
The valid range is 0 to 40 sec. The default value is 64*SIPT1Rtx
msec.
SIP destination port for sending initial SIP requests.
The valid range is 1 to 65534. The default port is 5060.
Note: SIP responses are sent to the port specified in the Via header.
Determines whether the 'user=phone' string is added to the SIP URI
and SIP To header.
[0] No = 'user=phone' string is not added.
[1] Yes = 'user=phone' string is part of the SIP URI and SIP To
header (default).
Determines whether the 'user=phone' string is added to the From
and Contact SIP headers.
[0] No = Doesn't add 'user=phone' string (default).
[1] Yes = 'user=phone' string is part of the From and Contact
headers.
Determines the format of the URI in the P-Asserted-Identity and P-
Preferred-Identity headers.
[0] Disable = 'sip:' (default).
[1] Enable = 'tel:'.
Defines the time (in seconds) that the device waits for a 200 OK
response from the called party (IP side) after sending an INVITE
message. If the timer expires, the call is released.
The valid range is 0 to 3600. The default value is 180.
305
6. Configuration Parameters Reference
Description
September 2009