AudioCodes Mediant 500 MSBR, Руководство пользователя

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Version 7.2 329 Mediant 500 MSBR
User's Manual 20. Control Network
Additional UDP Ports
additional-udp-
ports
[SIPInterface_Additional
UDPPorts]
Defines a port range for the <device>'s local, listening and source ports
for SIP signaling traffic over UDP. The parameter is used to assign a
specific local port to each SIP entity (e.g., PBX) communicating with a
common SIP entity (e.g., proxy server). This applies to the device's local
ports on the leg interfacing with the proxy server. In other words, this SIP
Interface is associated with the proxy server.
The valid range is 1,025 to 65535. The range is configured using the
syntax
x-y , where x is the starting port and y the ending port of the range
(e.g., 6000-7000). By default, the parameter is not configured.
For more information on configuring specific UDP ports for SIP entities
communicating with a common proxy server, see Configuring Specific
UDP Ports using Tag-based Routing on page 686.
Note:

The parameter's port range value must not overlap with the UDP port
configured by the 'UDP Port' parameter (SIPInterface_UDPPort). For
example, if the 'UDP Port' parameter is configured to 5070, you
cannot configure the 'Additional UDP Ports' parameter with a range of
5060-6000.

The parameter's port range value must not overlap with UDP port
ranges of Media Realms that are configured on the same network
interface. For example, if the RTP port range is 6000-6999, you must
configure the 'Additional UDP Ports' parameter to a range that is less
than 6000 or greater than 6999.

The maximum number of ports in the range is limited to the maximum
number of IP Groups that can be configured (see Configuring IP
Groups on page 333).

The parameter is applicable only to the SBC application.
Encapsulating Protocol
encapsulating-protocol
[SIPInterface_Encapsula
tingProtocol]
Defines the type of incoming traffic (SIP messages) expected on the SIP
Interface.

[0] No Encapsulation (default) = Regular (non-WebSocket) traffic.

[1] WebSocket = Traffic received on the SIP Interface is identified by
the device as WebSocket signaling traffic (encapsulated by
WebSocket frames). For outgoing traffic, the device encapsulates the
traffic using the WebSocket protocol (frames) on the TCP/TLS ports.
For more information on WebSocket, see SIP over WebSocket.
Note: WebSocket encapsulation is not supported for UDP ports.
Enable TCP Keepalive
tcp-keepalive-enable
[SIPInterface_TCPKeepa
liveEnable]
Enables the TCP Keep-Alive mechanism with the IP entity on this SIP
Interface. TCP keep-alive can be used, for example, to keep a NAT entry
open for clients located behind a NAT server, or simply to check that the
connection to the IP entity is available.

[0] Disable (default)

[1] Enable
Note: To configure TCP keepalive, use the following ini file parameters:
TCPKeepAliveTime, TCPKeepAliveInterval, and TCPKeepAliveRetry.
Used By Routing Server
used-by-routing-server
[SIPInterface_UsedByRo
utingServer]
Enables the SIP Interface to be used by a third-party routing server for
call routing decisions.

[0] Not Used (default)

[1] Used
For more information on the third-party routing server feature, see
Centralized Third-Party Routing Server on page 257.