Change SIP Connection page field descriptions

The default is Yes, which means the connection is available.

	Note:
	While you can configure multiple SIP connections, only one can be active at any one time.

The options are:

• TLS

• TCP

This must be a valid network address in the form of a fully qualified hostname or an IP address.

This field is enabled only when you select the Proxy Server option.

The default for TCP is 5060, and the default for TLS is 5061.

This field is enabled only when you select the Proxy Server option.

Calls are sent to the proxy server with the lowest priority value first. If this proxy server fails, calls are sent to the proxy server with the second lowest priority value. This continues up the proxy server list in priority order until either the call succeeds or the list is exhausted.

Enter a number in the range 0 to 65535. The default is 0.

This field is enabled only when you select the Proxy Server option.

For example, if proxy servers 1 and 2 are assigned a priority of 1, and weight of 4 and 6 respectively, then proxy server 1 has a 40% (4/(4+6)) chance of receiving a call while proxy server 2 has a 60% (6/(4+6)) chance.

Enter a number in the range 0 to 65535. The default is 0.

This field is enabled only when you select the Proxy Server option.

This field is enabled only when you select the Proxy Server option.

You cannot use the same proxy server address and port for adding another proxy server.

This field is enabled only when you select the Proxy Server option.

	Note:
	The DNS server must support the DNS SRV protocol.

The entry must be a valid hostname.

Ensure that the DNS server domain is configured to retrieve the ordered list of available server records which can be used to handle calls.

This field is enabled only when you select the DNS Server Domain option.

The default for TCP is 5060, and the default for TLS is 5061.

* (asterisk) means that all calls are routed to this trunk.

For Communications Manager, this should map to an extension configured on the switch.

Enter a number in the range 0 to 100. The default is 0.

Redirect attempt is disabled when the number in this field is set to 0.

The following options determine the SIP messages used for a VXML Consultative Transfer:

• INVITE with REPLACES: When Experience Portal receives INVITE with REPLACES in the SIP message, it establishes a secondary call to the transfer destination to:

  • Determine availability

  • Ensure that the destination answers within the established timeout

  The secondary call is then merged with the primary call that is being transferred. Experience Portal sends a request to the transferee for an INVITE message with a Replaces header that contains the information necessary to take control over the primary call. Experience Portal controls the progress of the call in case the response of the second call is not positive.

  	Note:
  	This option requires the transfer destination to support the INVITE with Replaces SIP message.

• REFER: With this option, the transferee determines the entire process of establishing the new call to the transfer destination. If the transfer destination is unavailable or does not respond to the call, the transferee sends the call to Experience Portal.

The default option is INVITE with REPLACES.

The options are:

• ASM (503): Sends a request to ASM to call another MPP that might be available.

• SES (480): Sends a request to SES to call another MPP that might be available.

• Custom: Allows custom response codes to be set for interoperability with other proxies and media gateways that might require a different response code to initiate a similar operation as stated above.

The default value is 250 millisecond(s).

Enter a number in the range of 10 to 8000 millisecond(s).

The default value is 2000 millisecond(s).

Enter a number in the range of 10 to 8000 millisecond(s).

The default value is 4000 millisecond(s).

Enter a number in the range of 500 to 180000 millisecond(s).

Enter a number from 1 to 99999.

• All Calls can be either inbound or outbound: This connection accepts any number of inbound or outbound calls up to the maximum number of calls defined in Maximum Simultaneous Calls.

• Configure number of inbound and outbound calls allowed: If this option is selected, Experience Portal displays the fields:

  • Inbound Calls Allowed: Enter the maximum number of simultaneous inbound calls allowed. This value must be less than or equal to the number of Maximum Simultaneous Calls.

  • Outbound Calls Allowed: Enter the maximum number of simultaneous outbound calls allowed. This value must be less than or equal to the number of Maximum Simultaneous Calls.

  The combined number of inbound and outbound calls must be equal to or greater than the number of Maximum Simultaneous Calls.

	Note:
	If all the SIP capacity configured (Maximum Simultaneous Calls) cannot be used because either the license or the total MPP capacity is not sufficient, then the number of inbound calls and outbound calls allowed will be reduced in proportion to the usable capacity.

• Yes: This connection uses SRTP.

• No: This connection does not use SRTP.

• AES_CM_128: This connection uses 128 key encryption.

• None: Messages sent through this connection are not encrypted.

• HMAC_SHA1_80: Authentication is done with HMAC SHA-1.

• HMAC_SHA1_32: Authentication is done with HMAC SHA-1.

• Yes: This connection uses RTCP encryption.

• No: This connection does not use RTCP encryption.

• Yes: This connection uses RTP authentication.

• No: This connection does not use RTP authentication.