The following section describes how to configure the WebLogic Network Gatekeeper and WebLogic SIP Server integration.
When using the WebLogic SIP server and WebLogic Network Gatekeeper together, the integration points between them must be configured in order for requests to be correlated between the SIP Server parts and Network Gatekeeper network tier. The WebLogic SIP Server part of the configuration is described in this section. The Network Gatekeeper configuration is described in the description of the individual communication service.
This is necessary when using the following network protocol plug-ins, since SIP Application sessions are used in the following communication services:
See BEA WebLogic SIP Server documentation for information on how to configure an manage Weblogic SIP Server.
There is an Network Gatekeeper Integration management console in SIP server, found under the node WLNG Integration under the Domain Structure in the SIP Server administration console.
Note: | The necessary integration software must be installed in the SIP server, see section Completing Post-Installation in the Network Gatekeeper Installation Guide. |
Configure the WebLogic SIP Server part of the Network Gatekeeper and WebLogic SIP Server integration:
This opens the WLNG Integration Configuration page.
Attribute: Suggested subscription lifetime
Attribute: Presence Server address
The other attributes should not be changed.
Below is a list of attributes and operations for configuration and maintenance of the general part of the Network Gatekeeper and SIP Server integration:
The URL to the JNDI provider in WebLogic Network Gatekeeper as configured during the domain configuration.
A WebLogic Network Gatekeeper administrative user.
The password associated with the WebLogic Network Gatekeeper administrative user.
Below is a list of attributes and operations for configuration and maintenance of the SIP Server part of the Parlay X 2.1 Third Party Call/SIP plug-in:
Class for that handles call management for Third Party call. Normally this field should not be edited.
The SIP servlet for Third Party Call. Normally this field should not be edited.
The Controller SIP URI that is used to establish the third party call. If this value is set, a call appears to the callee to come from this URI. By default, the value is “None”, where no controller URI will be used to establish the call. In this case, the call appears to the callee to come from the caller.
Below is a list of attributes and operations for configuration and maintenance of the SIP Server part of the Parlay X 2.1 Call Notification/SIP plug-in:
JNDI name for the Call Notification part of the Parlay X 2.1 Call Notification/SIP plug-in. Normally this field should not be edited.
JNDI name for the Call Direction part of the Parlay X 2.1 Call Notification/SIP plug-in. Normally this field should not be edited.
Below is a list of attributes and operations for configuration and maintenance of the SIP Server part of the Parlay X 2.1 Presence/SIP plug-in:
Specifies a suggested lifetime for a presence subscription, given in seconds.
This value might not be accepted by the Presence Server. In this case the Presence Server will set the expiry value it has chosen, and the value to use, in the first NOTIFY sent to the WebLogic SIP Server presence plug-in. The lifetime for the presence subscription will be according to the value received from the Presence Server.
The address to which the subscribe messages are sent. It can be the IP of the presence server or another IMS node that proxies the request.
The value is a SIP URI, for example sip:<host>:<port>