AudioCodes Mediant 1000 User Manual page 601

SIP User's Manual
Parameter
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]
Web: Use Tel URI for
Asserted Identity
[UseTelURIForAssertedID]
Web: Tel to IP No Answer
Timeout
EMS: IP Alert Timeout
[IPAlertTimeout]
Web: Enable Remote Party
ID
EMS: Enable RPI Header
[EnableRPIheader]
Web: Enable History-Info
Header
EMS: Enable History Info
[EnableHistoryInfo]
Version 6.4
used.
Persistent TCP connection ensures less network traffic due to fewer
setting up and tearing down of TCP connections and reduced latency
on subsequent requests due to avoidance of initial TCP handshake.
For TLS, persistent connection may reduce the number of costly TLS
handshakes to establish security associations, in addition to the initial
TCP connection set up.
Note: If the destination is a Proxy server, the TCP/TLS connection is
persistent regardless of the settings of this parameter.
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.
Defines the 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.
Enables Remote-Party-Identity headers for calling and called
numbers for Tel-to-IP calls.

[0] Disable (default).

[1] Enable = Remote-Party-Identity headers are generated in SIP
INVITE messages for both called and calling numbers.
Enables usage of the History-Info header.

[0] Disable (default)

[1] Enable
User Agent Client (UAC) Behavior:

Initial request: The History-Info header is equal to the Request-
URI. If a PSTN Redirect number is received, it is added as an
additional History-Info header with an appropriate reason.

Upon receiving the final failure response, the device copies the
History-Info as is, adds the reason of the failure response to the
601
A. Configuration Parameters Reference
Description
March 2012