AudioCodes E-SBC User Manual page 1211

CHAPTER 68 Configuration Parameters Reference
Parameter
configure voip >
sip-definition
settings >
reliable-conn-
persistent
[ReliableConnectionPersi
stentMode]
'TCP Timeout'
configure voip >
sip-definition
settings > tcp-
timeout
[SIPTCPTimeout]
'SIP Destination Port'
configure voip >
sip-definition
settings > sip-dst-
port
[SIPDestinationPort]
'Use user=phone in SIP
URL'
configure voip >
sip-definition
settings >
user=phone-in-url
[IsUserPhone]
'Use user=phone in From
Header'
configure voip >
sip-definition
settings > phone-
in-from-hdr
[IsUserPhoneInFrom]
'Use Tel URI for Asserted
Identity'
Mediant 1000 Gateway & E-SBC | User's Manual
■ [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 the
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 * 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.
Defines the format of the URI in the P-Asserted-Identity and P-
Preferred-Identity headers.
■ [0] Disable = (Default) 'sip:'
■ [1] Enable = 'tel:'
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Description