SIP line transport options

Option

Description

Default

Transport Protocol

Sets the transport protocol. Your selection depends on the protocols that are supported by your SIP-enabled devices (gateway, phones, and so on).

The options are:

• TCP (Transmission Control Protocol). The TCP station line is available if needed. Most new IP phones support TCP.

• TLS (Transport Layer Security or SSL). This option requires the Advanced Security feature license. After you select it, the TLS Security configuration option appears.

• UDP (User Datagram Protocol). Nearly all IP phones support UDP.

Note: If you change the transport protocol, you must deactivate and reactivate the line in order for the change to take effect. The line cannot be deactivated if any calls are active on it.

When a line is deactivated, no calls can be taken. Therefore, after you deactivate the line, reactivate it and then verify that it can take calls.

The other available options on this dialog box depend on the transport protocol that you select.

UDP

Audio Protocol

Indicates whether the audio stream is unencrypted or encrypted.

The options are:

• RTP (Real Time Protocol): The audio stream is unencrypted.

• SRTP (Secure RTP): The audio stream is encrypted.
  This option is available only if you select the TLS transport protocol. Select SRTP only if the endpoint(s) on this line support SRTP.
  If you select SRTP, the Security option is also available. Calls between devices that transmit and receive SIP TLS messages and SRTP audio are completely secure.

RTP (unencrypted)

Security

The Security setting determines, in part, whether the security lock icon appears in the CIC clients when a user places or receives an insecure call on this SIP line.  

The Security option is available only when you select the SRTP audio protocol.

In a CIC system environment, some devices may be configured to use SRTP while others do not. When two devices that use SRTP connect directly, both Interaction Clients display the lock icon to indicate that the call is secure from "end to end." The display of this lock icon is automatic and is not configurable.

If one device uses SRTP and another device does not, then at least one segment of a call between these devices is insecure. The audio between these devices needs to be transcrypted (converted) between SRTP and RTP via an intermediate device such as the media server.

If a SIP line handles insecure calls, you can configure the display of an open-lock icon to inform CIC client users that the call is not secure.

The options are:

• Minimal: hides the display of the open-lock icon on non-secure calls. If you select this option, completely secure calls always show the lock icon and all other calls show no lock icon. If a secure call creates a conference that includes a non-secure call, the lock icon disappears to indicate that the call is no longer secure.

• End-to-Edge: displays 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 (a gateway connected to the PSTN). It does not indicate security conditions on the PSTN or service provider outside of the CIC domain. If you select this option, secure calls always show the lock icon and all other calls that are non-secure show the open-lock icon.  If a secure call creates a conference that includes a non-secure call, all parties in the conference see the lock icon turn into an open-lock icon.  Conversely, if a non-secure conference call becomes secure from all of the end points to the edge of the CIC system, the open-lock icons change to lock icons.

Depends on your selection for the transport protocol

Adapter Name

Determines the local server IP address. The name of the adapter appears below the list.

This setting was formerly Address to Use.

The actual name of your network adapter.

This value is typically Local Area Connection because that is the default network adapter name that Windows uses.

Usable Addresses

CIC displays the list of IP addresses assigned for the chosen adaptor.

This option is available only if the EnableIPv6 server parameter is set to 1 (true).

Not specified.

Address Family

Sets the family of addresses to which CIC listens. CIC listens to all IP address assigned to each family.

This option is available only if the EnableIPv6 server parameter is set to 1 (true).

The options are:

• IPv4

• IPv6

• Telephony Default (IPv4 and IPv6). The host name resolves to both IPv4 and IPv6.

The default setting uses the order returned from a DNS query. If there are multiple IP addresses, CIC uses the first IP address in that range.

Media Address Family

Specifies the media address family to which CIC should listen.

This option is available only if the EnableIPv6 server parameter is set to 1 (true).

The options are

• IPv4

• IPv6

• Telephony Default (IPv4 and IPv6). The host name resolves to both IPv4 and IPv6. This option is typically used when CIC offers media, but it also helps to determine an answer when CIC receives identical media types and transport protocols for IPv4 and IPv6 information in the session description protocol (SDP).

Receive Port

UDP, TCP, and TLS: This option sets the port number for which the CIC SIP engine services requests.

The valid values are 1024 to 65535.

TLS runs on top of TCP. There is a conflict if TCP is set on the same port or the same protocol.

A new SIP line cannot have the same port and the same protocol as an existing SIP line. However, a new line may use the same port of an existing line if it uses a different protocol.

For more information, see PureConnect Security Features in the PureConnect  Documentation Library.

The default is 5060.

For TLS, this is set to 5061.

Connect Timer

TCP and TLS only: Sets the timer value in milliseconds for TCP connections on the SIP Line.  

The valid values are 500 to 20000 (milliseconds).

2000

T1 Timer (ms)

UDP only: Sets the timer value in milliseconds that represents the initial incremental delay between packet retransmission.

The valid values are 500 to T2 (milliseconds).

500

T2 Timer (ms)

UDP only: Sets the timer value in milliseconds that represents the maximum incremental delay between packet retransmissions.

The valid values are any values greater than or equal to 1000 (milliseconds).

1000

Maximum Packet Retry

UDP only: Sets the maximum number of packet retry attempts for requests.

Valid values are from 0 to 10.

4

Maximum Invite Retry

UDP only: Sets the maximum number of packet retry attempts for INVITE and ACK requests.

Valid values are from 0 to 6.

3

Reinvite Delay (ms)

UDP only: Sets the reinvite delay in milliseconds.

50

Retryable Reason Codes

Defines the list of valid SIP reason codes. If this line is part of a line group, and an outbound call that is made on this line returns a valid SIP reason code, then CIC retries the call on the next line in the line group.

Separate reason codes or ranges of reason codes with commas. For example:

"500-599"

Or...

"401, 480, 490-495, 500-599"

Note: "480" is not available on lines that are enabled for Microsoft Lync.

480, 500-599

Retryable Cause Codes

Defines a list of SIP cause codes. Cause codes take precedence over SIP response codes for retry attempts. If dial attempts are exhausted, the disconnect is treated as the most recent cause code if a cause code was present on any of the dial attempts. All dial attempts are traced at the note level when multiple retries are not treated as a 'no available lines' error.

Separate cause codes with commas. For example: "503, 507, 550"

The default value is 1-5,25,27,28,31,34,38,41,42,44,46,62,63,79,91,96,97,99,100,103

SIP DSCP Value

Sets the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) in transmitted SIP packets.

The values are shown in both hex (00..3F) and related decimal (0..63) formats. Some values are also identified by the binary format, CS6.

The range of valid values is 00 (0, 000000) through 3F (63, 111111).

18 (24, 011000) CS3

Inbound Progress Timer (ms)

Sets the number of milliseconds to wait before sending the 180 RINGING message. If the call is answered before this time expires, the 180 RINGING message is not sent.

Acceptable values are 1000 through 60,000 milliseconds

5000

No Inbound Progress Timer

Determines whether the 180 RINGING message is never sent on this SIP line.

Not selected

SIP Answer Delay (ms)

Sets the number of milliseconds of delay to insert before a call is answered. This setting is useful when there is some audio loss during call setup. The acceptable values are from 0 through 8,000 milliseconds. If the value is greater than or equal to 1000 milliseconds, an 180 Ringing SIP signal is sent before delay is inserted. Regardless of the value, the delay is always inserted after 200 OK is sent back.

500 milliseconds