SBC Parameters

Enables the Session Border Control (SBC) application.

Note:

Enables the device to terminate incoming in-dialog SIP OPTIONS messages or forward them to the outbound leg.

Determines whether incoming calls that cannot be classified (i.e. classification process fails) to a Source IP Group are rejected or processed.

Defines the maximum duration (in minutes) per SBC call (global). If the duration is reached, the device terminates the call.

The valid range is 0 to 35,791, where 0 is unlimited duration. The default is 0.

Note: You can also configure this feature per specific calls, using IP Profiles ('Max Call Duration' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles. If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Defines the timeout (in seconds) for SIP INVITE messages sent by the device (outbound IP routing).

The device starts the timeout when it sends the INVITE message and when (if) it receives the first SIP 18x response (e.g., 180 Ringing) from the called party. The timeout that is started when the INVITE message is sent, is only used if no 18x response is received.

If the timeout expires and no additional SIP response (for example, 200 OK) was received during this interval, the device releases the call.

The valid range is 0 to 3600 seconds. the default is 600.

Defines the maximum number of concurrent SIP SUBSCRIBE sessions permitted on the device.

The valid value is any value between 0 and the maximum supported SUBSCRIBE sessions. When set to -1, the device uses the default value. For more information, contact the sales representative of your purchased device.

Note: For the parameter to take effect, a device restart is required.

Enables the device to route in-dialog, refresh SIP SUBSCRIBE requests to the "working" (has connectivity) proxy.

Note: For this feature to be functional, ensure the following:

Defines which SIP SUBSCRIBE dialogs for registered users the device backs up, based on transport protocol. The parameter is applicable only when the device operates in HA mode.

Backing up SUBSCRIBE dialogs allows the redundant (now active) device to maintain user subscriptions and send relevant NOTIFY messages to the users (SIP UAs) after an HA switchover.

Note: For SUBSCRIBE dialogs that use TCP or TLS connections, it may be beneficial to configure the parameter to 1 or 2. For these registered users, the device needs to establish new TCP or TLS connections (typically initiated by the remote UA) after the switchover. Therefore, backing up these SUBSCRIBE dialogs is unnecessary, wasting valuable resources.

Enables the device to disconnect (delete from storage) SUBSCRIBE dialogs associated with registered users, upon an unregister, upon register expiration, or upon a refresh register done from a different source IP address / port (like when the transport protocol is TCP or TLS).

Defines the Max-Forwards SIP header value. The Max-Forwards header is used to limit the number of servers (such as proxies) that can forward the SIP request. The Max-Forwards value indicates the remaining number of times this request message is allowed to be forwarded. The count is decremented by each server that forwards the request.

The valid value range is 1-70. The default is 10.

The parameter affects the Max-Forwards header in the received message as follows:

Defines if the device connects or disconnects the call if it can't play the specified tone to the call party. This parameter relates to the feature that is described in Playing Tone upon Call Connect.

Enables the device to generate the 'tag' parameter's value in the SIP To header. This is applied to the first SIP response, received from the called party, which the device sends to the dialog-initiating SIP user agent (caller). In other words, this device-generated To tag overwrites the original To tag generated by the called party. All SIP messages between the device and caller use this generated To tag, while all SIP messages between the device and called party use the To tag generated by the called party. As the device-generated To tag value is short (up to 12 characters), this feature may be useful for SIP UAs that cannot handle long tag values.

An example of the To tag:

To: Alice@company.com; tag = 9777484849@10.10.1.110

Note: The feature is applicable only if the 'SBC Operation Mode' parameter is configured to B2BUA. This can be configured in the SRD and IP Groups table. However:

Defines the SBC session refresh timer (in seconds) in the Session-Expires header of outgoing INVITE messages.

The valid value range is 90 (according to RFC 4028) to 86400. The default is 180.

Defines the minimum amount of time (in seconds) between session refresh requests in a dialog before the session is considered timed out. This value is conveyed in the SIP Min-SE header.

The valid range is 0 (default) to 1,000,000, where 0 means that the device doesn't limit Session-Expires.

Defines the SIP user agent responsible for periodically sending refresh requests for established sessions (active calls). The session refresh allows SIP UAs or proxies to determine the status of the SIP session. When a session expires, the session is considered terminated by the UAs, regardless of whether a SIP BYE was sent by one of the UAs.

The SIP Session-Expires header conveys the lifetime of the session, which is sent in re-INVITE or UPDATE requests (session refresh requests). The 'refresher=' parameter in the Session-Expires header (sent in the initial INVITE or subsequent 2xx response) indicates who sends the session refresh requests. If the parameter contains the value 'uac', the device performs the refreshes; if the parameter contains the value 'uas', the remote proxy performs the refreshes. An example of the Session-Expires header is shown below:

Session-Expires: 4000;refresher=uac

Thus, the parameter is useful when a UA doesn't support session refresh requests or doesn't support the indication of who performs session refresh requests. In such a scenario, the device can be configured to perform the session refresh requests.

Note:

Defines additional time (in seconds) to add to the registration expiry time of users that are registered with the device.

The valid value is 0 to 15,500,000. The default is 0.

For more information, see Registration Refreshes.

Defines the method for searching a registered user in the device's User Registration database when a SIP INVITE message is received for routing to or from a user. If the registered user is found (i.e., destination URI in INVITE), the device routes the call to the user's corresponding contact address specified in the database.

Note: If the Request-URI contains the "tel:" URI or "user=phone" parameter, the device searches only for the user part.

Global parameter that defines the handling of the SIP P-Asserted-Identity header. You can also configure this feature per specific calls, using IP Profiles ('P-Asserted-Identity Header Mode' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Defines the device's handling of the SIP Contact header in REGISTER requests which it forwards as the outgoing message.

Note:

Defines which URI parameters are excluded when the device compares the URIs of two incoming dialog-initiating SIP requests (e.g., INVITEs) to determine if they were sent from a user that is registered in the device's registration database (registered AOR and corresponding Contact URI), during Classification.

The value of the parameter is a free-text string, which can't be empty. You can configure it to any sequence of parameters, separated by commas (e.g., "transport, maddr, ttl"). Alternatively, you can configure it to one of the following values (case-insensitive):

For example, if two SIP requests are received with different Contact header values, as shown below (in bold) and the parameter is configured to All, then the device considers these requests as received from the same registered user as it disregards the port (5060 and 5070), 'transport', and 'ttl' parameters in its comparison. If configured to None, the device considers these requests as received from two different registered users.

Contact: <sip:1000@172.17.142.105:5060;transport=tcp;ttl=10>

Contact: <sip:1000@172.17.142.105:5070;transport=tls;ttl=20>

Note: The AudioCodes proprietary "feu" string value for the user part must be included in the Contact header of REGISTER requests that the device forwards to the registrar server when the parameter is configured to a non-default value (i.e., not All). Therefore, if you configure the parameter to a non-default value, the [SBCKeepContactUserInRegister] parameter must not be configured to Keep User Without Unique Identifier (1).

Global parameter that defines the handling of SIP REFER requests. You can also configure this feature per specific calls, using IP Profiles ('Remote REFER Mode' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

When the SBCReferBehavior is set to 1, the device, while interworking the SIP REFER message, adds the prefix "T~&R-" to the user part of the URI in the Refer-To header. After this, the device can receive an INVITE with such a prefix (the INVITE is sent by the UA that receives the REFER message or 302 response). If the device receives an INVITE with such a prefix, it replaces the prefix with the value defined for the SBCXferPrefix parameter.

By default, no value is defined.

Note: This feature is also applicable to 3xx redirect responses. The device adds the prefix "T~&R-" to the URI user part in the Contact header if the SBC3xxBehavior parameter is set to 1.

Global parameter that defines the handling of SIP 3xx redirect responses. You can also configure this feature per specific calls, using IP Profiles ('Remote 3xx Mode' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Enables the device to include all previously negotiated media lines ('m=') within the current session in the SDP offer-answer exchange (RFC 3264).

For example, assume a call (audio) has been established between two endpoints and one endpoint wants to subsequently send an image in the same call session. If the parameter is enabled, the endpoint includes the previously negotiated media type (i.e., audio) with the new negotiated media type (i.e., image) in its SDP offer:

v=0
o=bob 2890844730 2890844731 IN IP4 host.example.com
s=
c=IN IP4 host.example.com
t=0 0
m=audio 0 RTP/AVP 0
m=image 12345 udptl t38

In this example, if the parameter is disabled, the only ‘m=’ line included in the SDP is the newly negotiated media (i.e., image).

Defines the URI type to use in the SIP Diversion header of the outgoing SIP message.

Note: The parameter is applicable only if the Diversion header is used. The [SBCDiversionMode] and [SBCHistoryInfoMode] parameters in the IP Profiles table determine the call redirection (diversion) SIP header to use - History-Info or Diversion.

Defines the cryptographic hash algorithm used in the outgoing authentication challenge (SIP 401 or 407) response when the device authenticates incoming SIP requests as an authentication server.

Defines if authentication of the SIP client is done locally (by the device) or by a RADIUS server.

Note:

Defines the lifetime (in seconds) that the current nonce is valid for server-based authentication. The device challenges a message that attempts to use a server nonce beyond this period. The parameter is used to provide replay protection (i.e., ensures that old communication streams are not used in replay attacks).

The valid value range is 30 to 600. The default is 300.

Defines the type of server-based authentication challenge.

Defines the authentication and integrity level of quality of protection (QoP) for digest authentication offered to the client. When the device challenges a SIP request (e.g., INVITE), it sends a SIP 401 response with the Proxy-Authenticate header or WWW-Authenticate header containing the 'qop' parameter. The QoP offered in the 401 response can be 'auth', 'auth-int', both 'auth' and 'auth-int', or the 'qop' parameter can be omitted from the 401 response. In response to the 401, the client needs to send the device another INVITE with the cryptographic hash algorithm (configured by [SIPServerDigestAlgorithm] parameter) of the INVITE message and indicate the selected auth type.

Global parameter that defines the duration (in seconds) of the periodic registrations that occur between the user and the device (the device responds with this value to the user). You can also configure this feature per specific calls, using IP Profiles ('User Registration Time' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Defines the duration (in seconds) for which the user is registered in the proxy database (after the device forwards the REGISTER message). This value is sent in the Expires header. When set to 0, the device sends the Expires header's value as received from the user to the proxy.

The valid range is 0 to 2,000,000 seconds. The default is 0.

Enables the device to change the expiry time in the Expires header of SIP 200 OK responses for user registration or subscription requests.

The feature is useful in scenarios where multiple users may refresh their registration or subscription simultaneously, causing the device to handle many such sessions at a given time. This may result in an overload of the device (reaching maximum session capacity), preventing the establishment of new calls or preventing the handling of some user registration or subscription requests. However, when this feature is enabled, the device assigns a random expiry time to each user registration or subscription, ensuring future user registration and subscription requests are more distributed over time (i.e., do not all occur simultaneously).

The valid value is 0 (disabled) to 20 (any value from 1 to 20 is considered enabled). The default is enabled (10). If disabled (i.e., 0), the device doesn't change the expiry time. If enabled, the device assigns a random expiry time as follows:

Note:

Defines the duration of the periodic registrations between the user and the device, when the device is in survivability state (i.e., when REGISTER requests cannot be forwarded to the proxy and are terminated by the device). When set to 0, the device uses the value set by the SBCUserRegistrationTime parameter for the device's response.

The valid range is 0 to 2,000,000 seconds. The default is 0.

Enables the device to notify Aastra IP phones that the device is currently operating in Survivability mode.

For more information, see Enabling Survivability Display on Aastra IP Phones.

Enables the interworking of dialog information (parsing of call identifiers in XML body) in SIP NOTIFY messages received from a remote application server.

For more information, see Interworking Dialog Information in SIP NOTIFY Messages.

Global parameter that enables the device to use the same call identification (SIP Call-ID header value) received in incoming messages for the call identification in outgoing messages. The call identification value is contained in the SIP Call-ID header.

You can also configure the feature per specific calls, using IP Profiles. For a detailed description of the parameter and for configuring the feature in the IP Profiles table, see Configuring IP Profiles.

Defines the handling of in-dialog SIP OPTIONS messages.

Defines the maximum duration (in seconds) that the device is prepared to wait for a response from external servers when a routing rule is configured to query an external server (e.g., LDAP server) on whose response the device uses to determine the routing destination.

The valid value is 0 to 60. The default is 10.

For more information, see Configuring a Routing Response Timeout.

Defines CPU cores allocation for optimizing a specific profile to achieve maximum capacity of that profile.

Note:

Defines the operational mode (transcoding or no transcoding) of the device's Media Component (MC).

For more information, refer to the Mediant Cloud Edition SBC Installation Manual.

Note: The parameter is applicable only to Mediant CE.

Defines Hyper-Threading capability on the device. Guests (i.e., the device) on VMware hypervisor do not inherit the Hyper-Threading capability of the host server. Therefore, this parameter allows you to override the hypervisor and define if Hyper-Threading is enabled or disabled on the device.

Note:

Determines the Globally Routable User Agent (UA) URI (GRUU) support, according to RFC 5627.

The parameter allows the device to act as a GRUU server for its SIP UA clients, providing them with public GRUU’s, according to RFC 5627. The public GRUU provided to the client is denoted in the SIP Contact header parameters, "pub-gruu". Public GRUU remains the same over registration expirations. On the other SBC leg communicating with the Proxy/Registrar, the device acts as a GRUU client.

The device creates a GRUU value for each of its registered clients, which is mapped to the GRUU value received from the Proxy server. In other words, the created GRUU value is only used between the device and its clients (endpoints).

Public-GRUU: sip:userA@domain.com;gr=unique-id

Enables authenticating a SIP BYE request before disconnecting the call. This feature prevents, for example, a scenario in which the SBC SIP client receives a BYE request from a third-party imposer assuming the identity of a participant in the call and as a consequence, the call between the first and second parties is inappropriately disconnected.

Enables the device to send a SIP 100 Trying response upon receipt of a SUBSCRIBE or NOTIFY message.

Enables the device to send a SIP 100 Trying response upon receipt of a re-INVITE request.

Defines the observer session expiry method when the IP Profile parameter, 'Session Expires Mode' is configured to Observer (see Configuring IP Profiles).

Enables SBC user registration for interoperability with BroadSoft's BroadWorks server, to provide call survivability in case of connectivity failure with the BroadWorks server.

Note: For a detailed description of this feature, see Enabling Auto-Provisioning of Subscriber-Specific Information of BroadWorks Server for Survivability.

Enables the Direct Media feature (i.e., no Media Anchoring) for all SBC calls, whereby SIP signaling is handled by the device without handling the RTP/SRTP (media) flow between the user agents (UA). The RTP packets do not traverse the device. Instead, the two SIP UAs establish a direct RTP/SRTP flow between one another. Signaling continues to traverse the device with minimal intermediation and involvement to enable certain SBC abilities such as routing

For more information on direct media calls, see Direct Media.

Global parameter that defines the voice transcoding mode (media negotiation). You can also configure this feature per specific calls, using IP Profiles ('Mediation Mode' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Defines the order of the Extension coders (coders added if there are no common coders between SDP offered coders and Allowed coders) and Allowed coders (configured in the Allowed Audio Coders Groups table) in the outgoing SIP message (in the SDP).

Note:

Enables the device to allocate DSP resources for a call at the SDP Offer or SDP Answer stage.

For more information on this feature, see Allocating DSPs on SDP Offer or Answer.

Global parameter that defines the handling of RTCP packets. You can also configure this feature per specific calls, using IP Profiles ('RTCP Mode' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Enables call forking of INVITE message received with a Request-URI of a specific contact registered in the device's database, to all users under the same AOR as the contact.

To configure call forking initiated by the device, see Initiating SIP Call Forking.

Enables the termination on the device of SIP REGISTER messages from secondary lines that belong to the Shared Line feature.

Note: The device always forwards REGISTER messages of the primary line.

Defines the handling of SIP 18x responses that are received due to call forking of an INVITE.

Enables synchronization of media between two SIP user agents when a call is established between them. Media synchronization means that the media is properly negotiated (SDP offer/answer) between the user agents. In some scenarios, the call is established despite the media not being synchronized. This may occur, for example, in call transfer (SIP REFER) where the media between the transfer target and transferee are not synchronized. The device performs media synchronization by sending a re-INVITE immediately after the call is established in order for the user agents to negotiate the media (SDP offer/answer).

Defines the SIP headers for which the device replaces “sips:” with “sip:” in the outgoing SIP-initiating dialog request (e.g., INVITE) when the destination transport type is unsecured (e.g., UDP). (The “sips:” URI scheme indicates secured transport, for example, TLS.)

Defines the duration (in seconds) for which the device attempts to detect fax (CNG tone) immediately upon the establishment of a voice session. The interval starts from the establishment of the voice call.

The valid value is 1 to any integer. The default is 10.

The feature applies to faxes that are sent immediately after the voice channel is established (i.e., after 200 OK).

You can configure the handling of fax negotiation by the device for specific calls, using IP Profiles ('Remote Renegotiate on Fax Detection' parameter). For more information, see Configuring IP Profiles.

Enables the device to overwrite the host part in SIP headers with IP addresses, unless the relevant host name parameters of the IP Group ('SIP Group Name' and 'SIP Source Host Name') are configured:

The parameter can be configured to one of the following values:

For more information on the IP Group parameters 'SIP Group Name' and 'SIP Source Host Name', see Configuring IP Groups.

Enables the rate-limiting Sliding Window Counter algorithm for Call Admission Control (CAC).

For more information, see Configuring Call Admission Control.

Enables Message Session Relay Protocol (MSRP).

For more information, see Configuring Message Session Relay Protocol.

Note: For the parameter to take effect, a device restart is required.

Defines the time (in seconds) before the user's registration with the device expires, at which the device sends an HTTP message to the Push Notification Server to trigger it into sending a push notification to the user to remind the user to send a REGISTER refresh message to the device.

The valid value range is 30 to 300. The default is 120.