AudioCodes Mediant 600 User Manual page 670

Parameter
sistentMode]
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
[UseTelURIForAssertedI
D]
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
User's Manual
dialog\transaction.

[1] = Enable - TCP connections to all destinations are persistent and
not released unless the device reaches 70% of its maximum TCP
resources.
While trying to send a SIP message connection, reuse policy
determines whether live connections to the specific destination are re-
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 is 64 multiplied by the
SipT1Rtx parameter value. For example, if SipT1Rtx is set to 500 msec,
then the default of SIPTCPTimeout is 32 sec.
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 = (Default) 'user=phone' string is part of the SIP URI and SIP
To header.
Determines whether the 'user=phone' string is added to the From and
Contact SIP headers.

[0] No = (Default) Doesn't add 'user=phone' string.

[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 = (Default) 'sip:'

[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 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)
670
Mediant 600 & Mediant 1000
Description
Document #: LTRT-83313