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lines.gifSIP 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

  • Delayed

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

  • Delayed

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.

You should always enable the Terminate Analysis on Connect option for a standalone fax station. Otherwise outbound faxing could fail.  

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

Configure a SIP line

PureConnect Customer Care site

Optional general server parameters

Media server general configuration

Lines concepts