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SIP Station Session

Use this page to configure your SIP station and/or the associated managed IP phone sessions.

Use Global SIP Station Session Settings (Station Configuration Only)

Select this checkbox to inherit the values defined at the Global SIP Station level.

Use SIP Session Timer and SIP Session Timeout

If checked, an OPTIONS messages will be sent every Sip Session Timeout (default is 60) seconds to the remote device. If the remote device does not respond to the OPTIONS message, the call will be disconnected.

SIP Register Interval

Select the amount of time in days, hours, minutes or seconds. The default value is 1 day.

Disconnect on Broken RTP

This parameter determines if a VoIP call will remain active after audio has been disrupted. Audio is considered disrupted if no RTP, RTCP and no comfort noise packet is received from the remote device. By default, this parameter is turned on (checked).

Media Timing

This is the timing on an INVITE request that contains a new media description in the SIP message body in the existing signaling session. Select Normal or Delayed from the pull-down list. Delayed is the default value for this setting.

See the latest version of SIP Application Note on the Product Information site. Select the Documentation link for the product and release you are using to open the PureConnect Documentation Library, then select the Telephony Application Notes link.

Media reINVITE Timing

This is the type of timing on a re-INVITE request that contains a new media description in the SIP message body in the existing signaling session. Select Normal or Delayed from the pull-down list. Delayed is the default value for this setting.

Terminate Analysis on Connect

Terminate Analysis on Connect is used to terminate the call analysis procedure when a SIP connection indication from the network is received. Select this box to enable this feature. The default is enabled.

For example, CIC makes its PSTN call via SIP calls through a SIP/ISDN gateway. This particular SIP/ISDN gateway only sends a SIP connect message back to CIC after the remote party answers the call. If call analysis is used, you would want to keep checked Terminate Analysis On Connect, so that call analysis terminates when the SIP connect message is received.

Another example, CIC makes its PSTN call via SIP calls through a SIP/analog gateway. This particular SIP/Analog gateway always sends a SIP connect message back to CIC prematurely, before the remote party answers the call. If call analysis is used, you would want to clear the Terminate Analyses On Connect checkbox, so that call analysis continues after the SIP connect message is received.

Tip: If the connection is to a station, the Terminate Analysis On Connect configured in the station is used.

Disable Media Server Passthru

Select this checkbox to this parameter to stop the media server from rewriting the SSRC header. This option is disabled (unchecked) by default.

Station Connections are Persistent

Select this box to maintain a persistent voice connection to the CIC server. The audio path will not disconnect until the station initiates the disconnection.

Clear this box to indicate when CIC determines that the audio path to the station is no longer needed, and CIC will initiate the disconnection.

Recommended setting

Operators: If you want to handle more calls than the phone is capable of handling. For example, if an operator wants to handle up to 20 simultaneous calls, then select this checkbox.

Call Center Agents: If call center agents are using an IP phone with a headset, and using a CIC client, this box should be selected.

Connection Call Warm Down Time

This value represents the number of seconds a connection call should remain connected after the regular call is disconnected. Once this timeout is expired, the connection call will be disconnected. The default value for this option is 5 seconds.

Note: This option is not used for persistent connection calls.

Call Appearances (does not apply to managed IP phones)

Select the number of call appearances the phone can handle. CIC will send up to the configured number of calls to the phone.

Note: If Persistent is selected, the number of call appearances will be 1.

Recommended setting

General: This value should be over 1 for experienced phone users only.

Cisco: The Cisco IP phone 7960 can have up to six line appearances (each line appearance is equivalent to a station). Each line appearance has a unique SIP address. Don't confuse line appearances with call appearances. Each line appearance handles 2 call appearances. Configure the phone to one line appearance and then this station configuration to 1 or 2 call appearances.

Pingtel: Pingtel Expressa IP phone has one line appearance that handles 4 call appearances. Set station configuration to 1, 2, 3, or 4 call appearances.