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SIP transport settings

The following table describes the SIP transport settings for the global SIP station, managed IP phones, and managed IP phone templates.

Setting

Description

Default  

Use Global SIP Station Transport Settings

 

This option specifies whether stations inherit the values that are defined for the global SIP station.

This option is available for station configurations only.

Selected

Use Proxy for Station Connections

 

This option indicates that the proxy list configured in the line configuration in Interaction Administrator should be used to connect stations.

Tip: If this option is not selected, CIC contacts the stations directly.

Not selected

Audio Protocol (does not apply to managed IP phones)

The audio stream on this SIP station can be unencrypted using RTP (Real Time Protocol) or encrypted using Secure RTP (SRTP). You must choose TLS as the Transport Protocol to use SRTP. Choose SRTP only if the endpoint(s) on this line support SRTP. If you select SRTP, it enables the Security option (below).  Calls between devices transmitting and receiving SIP TLS messages and SRTP audio are completely secure.

 

Security (does not apply to managed IP phones)

The Security list box is available only when you select SRTP as the Audio Protocol. The Security setting determines, in part, the visibility of the security icons on calls that appear in the CIC clients when placing or receiving calls via this SIP station.  

In an CIC system environment, some devices may support and be configured to use SRTP while other devices do not support SRTP or are not configured to use it.  When two devices (e.g., two stations) that support and are configured to use SRTP connect directly, both CIC clients will always display the lock icon because the call uses SRTP from one end to the other and is therefore secure.  This secure icon display is automatic and not configurable.

If one device supports and is configured to use SRTP and another device does not support or use SRTP, then at least one segment of a call between these devices is not secure. That means audio between these devices needs to be transcrypted (i.e., converted) between SRTP and RTP and vice versa via an intermediate device such as the Interaction Media Server.  SIP stations that handle calls that are not secure from one end to the other can use the Security list box to control the display of an open-lock icon to inform CIC client users that the call is not secure.

In the Security list box select Minimal to hide the display of the open-lock icon on non-secure calls.  In this case, completely secure calls will always show the lock icon and all other calls will show no lock icon.  If a secure call creates a conference and includes a non-secure call, the lock icon will disappear, indicating the call is no longer secure.

Select End-to-Edge to display the open-lock icon when a call, or at least one segment of a call in the CIC system domain is or becomes non-secure. End-to-edge means from one end of the call in the CIC system up to the edge of the CIC system (e.g., a gateway connected to the PSTN). It does not indicate security conditions on the PSTN or service provider outside of the CIC domain.  In this case, secure calls will always show the lock icon and all other calls that are non-secure will show the open-lock icon.  If a secure call creates a conference and includes a non-secure call, all parties in the conference will see the lock icon turn into an open-lock icon.  Conversely, if a non-secure conference call becomes secure from all the end points to the edge of the CIC system, the open-lock icons will change to lock icons.

 

Fax Protocol

Indicates the fax protocol to use.

 

The options are:

  • T30 only

  • T38 only

  • T38 then T30: CIC tries the T38 fax protocol first. If the recipient endpoint does not support this protocol, then CIC tries the T30 fax protocol.

  • T30 then T38: CIC tries the T30 fax protocol first. If the recipient endpoint does not support this protocol, then CIC tries the T38 fax protocol.

T38 only

SIP DSCP Value

 

This option indicates is the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) in transmitted SIP packets.

 

The available values appear in hex (00..3F) and related decimal (0..63) formats. Some values are also identified by the binary format, CS6. The range of values available is 00 (0, 000000) through 3F (63, 111111).  

18 (24, 011000) CS53

  

Related topics

Configure advanced options for managed IP phones and templates

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