AudioCodes Mediant 500 E-SBC, User Manual

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User's Manual 758 Document #: LTRT-10437
Mediant 500 E-SBC
Parameter Description
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.
Web/EMS: TCP Timeout
CLI: tcp-timeout
[SIPTCPTimeout]
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 * SipT1Rtx parameter
value. For example, if SipT1Rtx is set to 500 msec, then the default of
SIPTCPTimeout is 32 sec.
Web: SIP Destination Port
EMS: Destination Port
CLI: sip-dst-port
[SIPDestinationPort]
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.
Web: Use user=phone in
SIP URL
EMS: Is User Phone
CLI: user=phone-in-url
[IsUserPhone]
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.
Web: Use user=phone in
From Header
EMS: Is User Phone In
From
CLI: phone-in-from-hdr
[IsUserPhoneInFrom]
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.
Web: Use Tel URI for
Asserted Identity
CLI: uri-for-assert-id
[UseTelURIForAssertedID]
Determines the format of the URI in the P-Asserted-Identity and P-
Preferred-Identity headers.

[0] Disable = (Default) 'sip:'

[1] Enable = 'tel:'
Web: Tel to IP No Answer
Timeout
EMS: IP Alert Timeout
CLI: tel2ip-no-ans-timeout
[IPAlertTimeout]
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.
Web: Enable Remote Party
ID
EMS: Enable RPI Header
CLI: remote-party-id
[EnableRPIheader]
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.
Web: Enable History-Info
Header Enables usage of the History-Info header.

[0] Disable (default)