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Interaction Administrator Help
SIP line session options
The following table describes the options that you can use to configure session options for a SIP line. For information on how to access these options, see Configure a SIP line.
Changes to any options on this dialog box take effect immediately.
Option |
Description |
Default |
Use SIP Session Timer
|
Determines whether a recurring OPTIONS message is sent to the remote device. If the remote device does not respond to the OPTIONS message, the call is disconnected. |
Not selected |
SIP Session Timeout |
The recurrence interval of the OPTIONS message that is sent when the Use SIP Session Timer check box is selected. |
60 seconds |
Disconnect on Broken RTP
|
Determines if a VoIP call remains active if the audio is disrupted. Audio is considered disrupted if no RTP, RTCP, and no comfort noise packet is received from the remote device. |
Not selected |
Media Timing
|
The options are:
|
Normal |
Media reINVITE Timing
|
Specifies if the media attempts to add media streams to the session immediately or if the timing is delayed when it receives a reINVITE. The options are:
|
Normal |
Terminate Analysis on Connect
|
Terminates the call analysis procedure when a SIP connection indication from the network is received. For example, Interaction Center makes its PSTN call via SIP calls through a SIP/ISDN gateway. In this example, the SIP/ISDN gateway sends only a SIP connect message back to Interaction Center after the remote party answers the call. If call analysis is used, you would want to select Terminate Analysis On Connect, so that call analysis terminates when the SIP connect message is received.
For example, Interaction Center makes its PSTN call via SIP calls through a SIP/Analog gateway. In this example, the SIP/Analog gateway always sends a SIP connect message back to Interaction Center prematurely, before the remote party answers the call. If call analysis is used, you would want to deselect Terminate Analysis On Connect, so that call analysis continues after the SIP connect message is received.
If the connection is to a station, the Terminate Analysis On Connect configured in the station is used. Notes:
Beginning in IC 4.0 SU3, CIC terminates call analysis when it
connects to remote stations.
This decreases the time it takes to connect the caller to the
agent at the remote station. If you want to restore the previous
functionality of using call analysis when connecting to remote
stations, create the Remote Station Call Analysis Answer Supervision
Interaction Center server parameter, and set it to False. |
Not selected |
Disable Media Server Passthru
|
Determines whether the media server rewrites the SSRC header. |
Not selected |
ASR Enabled
|
Determines whether ASR (Automatic Speech Recognition) resources are allocated for this SIP line. |
Selected |
Related topics
PureConnect Customer Care site
Optional general server parameters