To create or modify an existing SIP Profile:
- In the WebUI, click the Settings tab.
In the left navigation pane, go to SIP > SIP Profiles.
Figure : SIP Profile Table
Modifying a SIP Profile
- Click the expand ( ) Icon next to the entry you wish to modify.
- Edit the entry properties as required, see details below.
Creating a SIP Profile
Click the Create SIP Profile (
) icon at the top of the SIP Profile View page.
Figure : Create SIP Profile
Session Timer - Field Definitions
The options in the Session Timer panel control the timers used to verify that the current SIP session is still valid.
Specifies whether or not to use the session timer to verify the SIP session. The remainder of the options in this panel are only visible of the Session Timer is Enabled (default).
Minimum Acceptable Timer
Specifies the minimum value to use for incoming session refresh in seconds. If the SBC Edge (SBC) negotiates a session refresh this value will be used to indicate the minimum value acceptable for a session refresh. Default: 600.
Offered Session Timer
Specifies an interval, proposed to a SIP peer, to use for a session refresh. At each interval, a session refresh occurs, if the remote does not respond the call is disconnected. Default: 3600.
Terminate on Refresh Failure
Controls how the SBC responds if the session refresher fails. Default: False.
- When SBC is the refresher, the configuration item, ‘Terminate on refresh failure’ does not apply, That is, anytime the session timer fires the SBC will send either a re-Invite or Update.
- When Peer is the refresher (Not SBC), the following applies:
a. If ‘Terminate on refresh failure = false’, SBC will send a re-Invite or Update on session timer fire.
b. If ‘Terminate on refresh failure = true’, SBC will terminate the call by sending a BYE when the session timer fires.
- Set the timer value is as follows:
- Set timer X seconds before the session expires, where X is the smaller value of 32 or 1/3 of the session timer value.
Example: Session timer = 180, then X=32 since 32 is less than 60. Then, timer will be set for 180-32 = 148 seconds.
Header Customization - Field Definitions
The options in the Header Customization panel determine how certain headers in the SIP protocol are handled, what content from certain headers is used, and more.
UA (User Agent) Header
Specifies the contents of the SIP User-Agent header. Default: UX.
FQDN in From Header
Specifies which, if any, Fully Qualified Domain Name (FQDN) is used in the From header.
- Disable: SBC IP address is used in the From Header. (Default)
- Sonus SBC FQDN: SBC FQDN is used in the From Header.
- Server FQDN: Next hop SIP Server (IP address or FQDN depending of SIP server table entry) is used in the From Header.
- Static: Host portion (FQDN or IP) to be used in the From Header. If Static is selected, you must configure the host portion through the Static Host field.
FQDN in From Header will not function if the Host Name and/or Domain Name is not configured in System | Node-Level Settings.
Determines (Enable/Disable) whether FQDN (Fully Qualified Domain Name) is included in the Contact Header. Default: Disable.
Specifies the host portion to be used in the From Header. This field is displayed only when the FQDN in From Header field is set as Static. Valid entry: FQDN[:port], IP4[:port], [IP6]:port or IP6. An example of an IPv6 entry is: [fd00:10:6b56:226::204]:27526.
Send Assert Header
When enabled (set as Always), the SBC always sends a P-Asserted-Identity header in the outbound INVITE message. When the SBC receives an inbound INVITE message with a P-Preferred-Identity or P-Asserted-Identity header, the information from that header is used in the outbound P-Asserted-Identity header. However, when the SBC receives a Privacy header, the identity is revealed in the outbound INVITE based on privacy header value.
When disabled, privacy information in the outbound INVITE is sent depending on the configuration of the Trusted Interface and the Privacy Pass-through Header. Default: Trusted Only.
Sonus Diagnostics Header
Specifies if the X-Sonus-Diagnostics header is added to the outbound SIP signaling messages. This header indicates detailed diagnostic information about the call (such as media mode, DSP/RTP-Proxy/Direct etc.).
Options: Enable (default)(header is included), or Disable (header is not included).
Example: In a Direct Media scenario, the SBC is only anchoring SIP Signaling and the media failures are outside of the control of SBC. In this case, the following header message is included:
X-Sonus-Diagnostics: SBCInternal;cid=124;media-mode="audio:Direct video:Direct".
This header tells the peer device that the SBC has used direct media mode.
When enabled (default), the SBC passes the Display Name (Calling Name), P-Asserted-Identity and Privacy header information from the inbound INVITE to the outbound INVITE message. The User-Agent header is also present.
When disabled, the Display Name (Calling Name), PAI/PPI and Privacy headers are not passed, and the User-Agent header is suppressed.
Calling Info Source
Enables SIP to use Calling Number and Name from FROM header always. By default, if P-Asserted-ID/P-Preferred-ID are present, Calling Number and Name are derived from P-Asserted-ID/P-Preferred-ID headers. Valid entry: From RFC Standard (default) or "From" Header only.
Diversion Header Selection
Determines which SIP Header to use in SIP call. Valid entry: First or Last (default).
Record Route Header
Determines which technical specification needs to be used to process Record-Route header. Choices: ETSI or RFC 3261(default).
Timers - Field Definitions
The options in this panel determine the duration of timers that control the behavior or SIP transmissions.
Transport Timeout Timer
Specifies the interval, in milliseconds, for which new TCP/TLS connections over SIP can be tried when no response is received from far end. Default: 5000.
Specifies the maximum number of retransmissions of client transaction messages over UDP. The default value disables this feature and maximum retransmissions are then controlled by Timer B and Timer F values. Any non-zero value modifies Timer B and Timer F and retransmissions are controlled by the selected value. Default: RFC Standard.
Enter an integer value for each of the timers, if other than standard defaults are required.
Round-trip time (RTT) estimate
Maximum retransmission interval for non-INVITE requests and INVITE responses
Maximum duration that a message can remain in the network
Maximum time to wait when the server transaction enters the "Completed" state.
Wait time for response retransmissions
Transport Timeout Timer
Determines the duration of time to wait before ending a tired connection when far end blocks the connection.
MIME Payload - Field Definitions
The MIME Payload contains location information about the origin of the call. The SBC processes the Multi-part MIME Content-Type header in the SIP INVITE message upon ingress to the Signaling Group. The SBC passes through the pidf+xml Content-Type as is to the egress side Signaling Group in an SIP INVITE message. The SBC does not process the XML body of the PDIF-LO content.
ELIN identifier maps the selected label from PIDF-LO XML content to the call router translation table input field. Default value: LOC.
- LOC: Location information (Generic information, i.e., city or building)
- FLR: Floor location
- HNO: House number
Specifies whether or not to allow the passthrough of pidf+xml content from Ingress to Egress.
- PIDF-LO: GEOPRIV Presence Information Data Format Location Object (PIDF-LO) The Presence Information Data Format Location Object (PIDF-LO) is the recommended way of encoding location information and associated privacy policies. Location information in a PIDF-LO may be described in a geospatial manner based on a subset of Geography Markup Language (GML) or as civic location information. Default value: Enable.
Unknown Subtype Passthrough
Specifies whether or not to allow the passthrough of unknown subtype content from Ingress to Egress. Default value: Disable.
The SIP Option Tags determine in which SIP header, the 100rel (Default: Supported), Path (Default: Not Present), Timer (Default: Supported) and Update (Default: Supported) tags will appear.
- Supported: means the specified tag is included in the Supported Header.
- Required: means the specified tag is included in the Required Header.
- Not Present: means the specified tag is included in neither the Supported Header nor the Required Header.
When the Session Timer option is enabled, the SBC uses either INVITE or UPDATE to refresh the session. The method used is negotiated between the SBC and peer. The Option Tags selections simply specify in which header to specify the tag.
The SDP customization fields enables you to customize configuration for SDP. The features in this section are applicable only when a DSP is in use for the call.
Send Number of Audio Channels
For interoperability purposes, enables the /1 field to be removed from media attribute in SDP. In the a= line, the /1 field is dropped if the value is configured as False.
Example: a=rtpmap:4 g723/8000/1 becomes a=rtpmap:4 g723/8000
Valid entry: True (/1 field stays intact) or False (/1 field is dropped). Default value: False.
Determines if the Connection Data (represented by "c=") is present in both the Session and Media sections of the SDP. Some endpoints are unable to handle the "c=" in the media section. Valid entries: True (the "c=" is included in the media section) or False (the "c=" does not appear in the media section). Default entry: True.
Origin Field Username
Value of the Origin (o=) username field. Valid entry: up to 64 alphanumeric characters. For security purposes, this field can be set as "-". Default value: SBC.
Value of the Session (s=) field. Valid entry: up to 64 alphanumeric characters. For security purposes, this field can be set as "-". Default value: VoipCall.
Digit Transmission Preference
Determines how digits are sent to the far end during an ongoing call (Example, digits might be sent to an IVR system during a call). This argument specifies which digit transmission method to use when both are available to the call.
- RFC 2833/Voice (default)
- SIP Info
SDP Handling Preference
Controls the way the SBC generates the Offer. The Answer generated by the SBC is determined according to the peer's Officer, and whether it follows RFC 3264. Two options are available:
- Legacy Audio/Fax. Only one media line is used while sending out Offer for Audio or T.38 switching.
- RFC 3264. Multiple media lines are used as per RFC 3264 while sending out the Offer.