AudioCodes Mediant 500 MSBR, Руководство пользователя

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Version 7.2 1221 Mediant 500 MSBR
User's Manual 74. Configuration Parameters Reference
settings > sbc-forking-
handling-mode
[SBCForkingHandlingMo
de]
INVITE-initiating UA. If SIP 18x with SDP is received, the device
opens a voice stream according to the received SDP and disregards
any subsequent 18x forking responses (with or without SDP). If the
first response is 180 without SDP, the device sends it to the other
side.

[1] Sequential = All 18x responses are forwarded, one at a time
(sequentially) to the INVITE-initiating UA. If a 18x arrives with an
offer only, then only the first offer is forwarded to the INVITE-
initiating UA and subsequent 18x responses are discarded.
Gateway Direct Route
Prefix
configure voip > sbc
settings > gw-direct-
route-prefix
[GWDirectRoutePrefix]
Defines the prefix destination Request-URI user part that is appended
to the original user part for alternative IP-to-IP call routing from SBC to
Gateway (Tel) interfaces.
The valid value is a string of up to 16 characters. The default is
"acgateway-<original prefix destination number>". For example,
"acgateway-200".
For more information, see Configuring SBC IP-to-IP Routing Rules on
page 671.
configure voip > sbc
settings > sbc-media-
sync
[EnableSBCMediaSync]
Enables synchronization of media between two SIP user agents when
a call is established between them. Media synchronization means that
the media is properly negotiated (SDP offer/answer) between the user
agents. In some scenarios, the call is established despite the media not
being synchronized. This may occur, for example, in call transfer (SIP
REFER) where the media between the transfer target and transferee
are not synchronized. The device performs media synchronization by
sending a re-INVITE immediately after the call is established in order
for the user agents to negotiate the media (SDP offer/answer).

[0] Disable = (Default) Media synchronization is performed only if
the RTP mode (e.g., a=sendrecv, a=sendrecv, a=sendonly,
a=recvonly, and a=inactive) between the user agents are different
and synchronization is required.

[1] Enable = Media synchronization is performed if the media,
including RTP mode or any other media such as coders, is different
and has not been negotiated between the user agents.

[2] Never = Media synchronization is never performed.
[SBCRemoveSIPSFrom
NonSecuredTransport]
configure voip >
sbc settings >
sbc-remove-sips-
non-sec-transp
Defines the SIP headers for which the device replaces “sips:” with “sip:”
in the outgoing SIP-initiating dialog request (e.g., INVITE) when the
destination transport type is unsecured (e.g., UDP). (The “sips:” URI
scheme indicates secured transport, for example, TLS.)

[0] = (Default) The device replaces “sips:” with “sip:” for the
Request-URI and Contact headers only (and retains “sips:” for all
other headers).

[1] = The device replaces “sips:” with “sip:” for the Request-URI,
Contact, From, To, P-Asserted, P-Preferred, and Route headers.
Admission Control Table
Admission Control
configure voip > sbc sbc-
admission-control
[SBCAdmissionControl]
Defines Call Admission Control (CAC) rules.
The format of the ini file table parameter is as follows:
[SBCAdmissionControl]
FORMAT SBCAdmissionControl_Index =
SBCAdmissionControl_AdmissionControlName,
SBCAdmissionControl_LimitType,
SBCAdmissionControl_IPGroupName,
SBCAdmissionControl_SRDName,
SBCAdmissionControl_SIPInterfaceName,