AudioCodes Mediant 800B - Page 1131

Version 7.2 1131 Mediant 800B Gateway & E-SBC
User's Manual 72. Configuration Parameters Reference
Parameter Description
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 the parameter.
TCP Timeout
configure voip > sip-
definition settings > 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.
SIP Destination Port
configure voip > sip-
definition settings > 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.
Use user=phone in SIP URL
configure voip > sip-
definition settings >
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.
Use user=phone in From
Header
configure voip > sip-
definition settings > 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.
Use Tel URI for Asserted
Identity
configure voip > sip-
definition settings > 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:'
Tel to IP No Answer
Timeout
configure voip > gateway
advanced > 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, for Tel-to-IP calls. If the timer expires, the call is released.
The valid range is 0 to 3600. The default is 180.
Enable Remote Party ID
configure voip > sip-
definition settings > 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.
Enable History-Info Header Enables usage of the SIP History-Info header.