AudioCodes Mediant 4000 SBC, Руководство пользователя

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CHAPTER 60 Configuration Parameters Reference
Mediant 4000 SBC | User's Manual
Parameter Description
Note: To enable TCP/TLS connection re-use, set the
EnableTCPConnectionReuse parameter to 1.
'Reliable Connection
Persistent Mode'
configure voip >
sip-definition
settings >
reliable-conn-
persistent
[ReliableConnectionP
ersistentMode]
Enables setting of all TCP/TLS connections as persistent and
therefore, not released.
■ [0] Disable = (Default) All TCP connections (except those that
are set to a proxy IP) are released if not used by any SIP
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 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' 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.
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