SBC Parameters

The SBC and CRP parameters are described in the table below.

SBC and CRP Parameters

Parameter

Description

CRP-specific Parameters

configure voip > application > enable-crp

[EnableCRPApplication]

Enables the CRP application.

[0] = Disable (default)
[1] = Enable

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

'CRP Survivability Mode'

configure voip > sbc settings > crp-survivability-mode

[CRPSurvivabilityMode]

Defines the CRP mode.

[0] Standard Mode (default)
[1] Always Emergency Mode
[2] Auto-answer REGISTER

'CRP Gateway Fallback'

configure voip > sbc settings > crp-gw-fallback

[CRPGatewayFallback]

Enables fallback routing from the proxy server to the Gateway (PSTN).

[0] Disable (default)
[1] Enable

SBC-specific Parameters

configure voip > application > enable-sbc

[EnableSBCApplication]

Enables the Session Border Control (SBC) application.

[0] = Disable
[1] = Enable (default)

Note:

For the parameter to take effect, a device restart is required.
The parameter is enabled by default only if the License Key contains at least one of the SBC-related capacity features (e.g., "SBC-Signaling"); otherwise, the parameter is disabled.

SBC and CRP Parameters

'Terminate Inbound OPTIONS'

configure voip > sbc settings > sbc-terminate-options

[SBCTerminateOptions]

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

[0] Disable
[1] Enable (default)

'Unclassified Calls'

configure voip > sbc settings > unclassified-calls

[AllowUnclassifiedCalls]

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

[0] Reject = (Default) Call is rejected if classification fails.
[1] Allow = If classification fails, the incoming packet is assigned to a source IP Group (and subsequently processed) as follows:
The source SRD is determined according to the SIP Interface to where the SIP-initiating dialog request is sent. The source IP Group is set to the default IP Group associated with this SRD.
If the source SRD is ID 0, then source IP Group ID 0 is chosen. In case of any other SRD, then the first IP Group associated with this SRD is chosen as the source IP Group or the call. If no IP Group is associated with this SRD, the call is rejected.

'SBC Max Call Duration'

configure voip > sbc settings > sbc-mx-call-duration

[SBCMaxCallDuration]

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.

'SBC No Answer Timeout'

configure voip > sbc settings > sbc-no-alert-timeout

[SBCAlertTimeout]

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.

configure voip > sbc settings > num-of-subscribes

[NumOfSubscribes]

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.

configure voip > sbc settings > sbc-dialog-subsc-route-mode

[SBCInDialogSubscribeRouteMode]

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

[0] = (Default) Disable – the device sends in-dialog, refresh SUBSCRIBES according to the address in the Contact header of the 200 OK response received from the proxy to which the initial SUBSCRIBE was sent (as per the SIP standard).
[1] = Enable – the device routes in-dialog, refresh SUBSCRIBES to the "working" proxy (regardless of the Contact header). The "working" proxy (address) is determined by the device's keep-alive mechanism for the Proxy Set that was used to route the initial SUBSCRIBE.

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

Keep-alive mechanism is enabled for the Proxy Set ('Proxy Keep-Alive' parameter is set to any value other than Disable).
Load-balancing between proxies is disabled ('Proxy Load Balancing Method' parameter is set to Disable).

config voip > sbc settings > backup-subscriptions

[BackupSubscriptions]

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.

[0] = (Default) The device backs up all SUBSCRIBE dialogs (regardless of used transport protocol).
[1] = The device backs up only SUBSCRIBE dialogs that use UDP.
[2] = The device doesn't back up any SUBSCRIBE dialogs.

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.

config voip > sbc settings > disconnect-subscriptions

[DisconnectSubscriptionsMode]

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).

[0] = (Default) Disable - the device stores the SUBSCRIBE dialogs of registered users until the subscriptions' expiration times are reached.
[1] = Enable - the device disconnects (deletes) SUBSCRIBE dialogs of registered users upon an unregister, upon register expiration, or upon a refresh register done from a different source IP address / port.

Note: The parameter is applicable only to the SBC application.

configure voip > sbc settings > sbc-max-fwd-limit

[SBCMaxForwardsLimit]

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:

If the received header’s original value is 0, the message is not passed on and is rejected.
If the received header’s original value is less than the parameter's value, the header’s value is decremented before being sent on.
If the received header’s original value is greater than the parameter's value, the header’s value is replaced by the user-defined parameter’s value.

'Play Tone on Connect Failure Behavior'

play-tone-on-connect-failure-behavior

[PlayToneOnConnectFailureBehavior]

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.

[0] Disconnect (Default)
[1] Ignore

configure voip > sip-definition settings > force-generate-to-tag

[ForceGenerateToTag]

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

[0] = Disable (default). The device forwards the To tag transparently between the SIP UAs.
[1] = Enable. The device generates the To tag in the response sent to the initiator of the SIP dialog.

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:

The IP Group's 'SBC Operation Mode' parameter takes precedence over the SRD's 'SBC Operation Mode' parameter. For example, if the IP Group is configured for B2BUA but its' associated SRD is not, then the tag-generation feature can function.
If the IP Group's 'SBC Operation Mode' parameter is not configured (-1), the tag-generation feature for the IP Group is functional only if its' associated SRD is configured for B2BUA.
For call routing between IP Groups, the feature can only function if both IP Groups are configured for B2BUA, or if one or both of them is not configured (-1), but the associated SRD is configured for B2BUA.

'Session-Expires'

configure voip > sbc settings > sbc-sess-exp-time

[SBCSessionExpires]

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.

Note: The parameter is applicable only to the SBC application.

'Minimum Session-Expires'

configure voip > sbc settings > min-session-expires

[SBCMinSE]

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.

Note: The parameter is applicable only to the SBC application.

configure voip > sbc settings > sbc-session-refresh-policy

[SBCSessionRefreshingPolicy]

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.

[0] = (Default) Remote Refresher. The UA (proxy) performs the session refresh requests. The device indicates this to the UA by sending the SIP message with the 'refresher=' parameter in the Session-Expires header set to 'uas'.
[1] = SBC Refresher. The device performs the session refresh requests. The device indicates this to the UA by sending the SIP message with the 'refresher=' parameter in the Session-Expires header set to 'uac'.

Note:

The time values of the Session-Expires (session refresh interval) and Min-SE (minimum session refresh interval) headers can be configured using the [SBCSessionExpires] and [SBCMinSE] parameters, respectively.

Note: The parameter is applicable only to the SBC application.

'User Registration Grace Time'

configure voip > sbc settings > sbc-usr-reg-grace-time

[SBCUserRegistrationGraceTime]

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.

'DB Routing Search Mode'

configure voip > sbc settings > sbc-db-route-mode

[SBCDBRoutingSearchMode]

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.

[0] All permutations = (Default)
To User: Device searches for the user in the database using the entire Request-URI (user@host). If not found, it searches for the user part of the Request-URI. For example, it first searches for "4709@joe.company.com" and if not found, it searches for "4709".
From User: Device searches for the user in the database using the entire From header AOR (user@host). If not found, it searches for the user part of the From header AOR. For example, it first searches for "4709@domain.com" and if not found, it searches for "4709".
[1] Dest URI dependant =
To User: Device searches for the user in the database using the entire Request-URI (user@host) only. For example, it searches for "4709@joe.company.com".
From User: Device searches for the user in the database using the entire From header AOR (user@host) only. For example, for “From: <sip:4709@domain.com>", the device searches for “4709@domain.com”.

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

'Handle P-Asserted-Identity'

configure voip > sbc settings > p-assert-id

[SBCAssertIdentity]

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.

'Keep original user in Register'

configure voip > sbc settings > keep-contact-user-in-reg

[SBCKeepContactUserinRegister]

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

[0] Do not keep user; override with unique identifier = (Default) The device replaces the user part of the Contact header with a unique value, for example:
Incoming Contact Header: <sip:123@domain.com>
Outgoing Contact Header: <sip:FEU1-7-1-3@SBC>
[1] keep user without unique identifier = The device retains the original user part value of the Contact header in the outgoing REGISTER request.
[2] Keep user; add unique identifier as URI parameter = The device retains the original user part value of the Contact header in the outgoing REGISTER request. In addition, it adds the special URI parameter "ac-feu=<identifier>" to the Contact header, which is used to differentiate between two SIP entities with the same user part. The identifier value is generated by the device.
Incoming Contact Header: <sip:123@domain.com>
Outgoing Contact Header: <sip:123@SBC;ac-feu=1-7-1-3>

Note:

The parameter is applicable only to REGISTER messages received from User-type IP Groups which are sent to Server-type IP Groups.
Depending on the 'Remote Representation Mode' parameter of the IP Profiles table ('Remote Representation Mode' parameter), the host part in the SIP Contact header can replaced by the device’s IP address or by the value of the 'SIP Group Name' parameter (configured in the IP Groups table).

'URI Comparison Excluded Parameters'

config-voip > sbc settings > uri-comparison-excluded-params

[SBCURIComparisonExcludedParams]

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):

All = (Default) All URI parameters (except "gr" (gruu), "user=phone", and AudioCodes proprietary "ac-int" parameters) and ports are excluded in the comparison of the two URIs. Therefore, if there are two different registrations of the same user whose Contacts are differentiated only by ports and/or a proprietary parameter, the device considers them to be the same single registration, even though they are different registrations.
None = All URI parameters and ports are included in the comparison of the two URIs.
Port = The ports of the URIs are excluded in the comparison of the two URIs, but all other URI parameters are included in the comparison. "port" can be combined with other URI parameters that you want excluded (e.g., port, transport, proprietary-param).

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).

'SBC REFER Behavior'

configure voip > sbc settings > sbc-refer-bhvr

[SBCReferBehavior]

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.

configure voip > sbc settings > sbc-xfer-prefix

[SBCXferPrefix]

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.

configure voip > sbc settings > sbc-3xx-bhvt

[SBC3xxBehavior]

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.

'Enforce Media Order'

configure voip > sbc settings > enforce-media-order

[SBCEnforceMediaOrder]

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

[0] Disable (default)
[1] Enable

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).

'SBC Diversion URI Type'

configure voip > sbc settings > sbc-diversion-uri-type

[SBCDiversionUriType]

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

[0] Transparent = (Default) The device doesn't change the URI and leaves it as is.
[1] Sip = The "sip" URI is used.
[2] Tel = The "tel" URI is used.

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.

configure voip > sbc general-setting > sip-server-digest-algorithm

[SIPServerDigestAlgorithm]

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.

MD5 (default)
SHA-256

configure voip > sbc settings > sbc-server-auth-mode

[SBCServerAuthMode]

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

[0] = (Default) Authentication is done by the device (locally).
[1] = Authentication is done by an RFC 5090 compliant RADIUS server.
[2] = Authentication is done according to the Draft Sterman-aaa-sip-01 method.

Note:

Currently, option [1] is not supported.
The parameter is overridden by the IP Group parameter 'SBC Server Authentication Type'.

'Lifetime of nonce'

configure voip > sbc settings > lifetime-of-nonce

[AuthNonceDuration]

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.

'Authentication Challenge Method'

configure voip > sbc settings > auth-chlng-mthd

[AuthChallengeMethod]

Defines the type of server-based authentication challenge.

[0] 0 = (Default) Send SIP 401 "Unauthorized" with a WWW-Authenticate header as the authentication challenge response.
[1] 1 = Send SIP 407 "Proxy Authentication Required" with a Proxy-Authenticate header as the authentication challenge response.

'Authentication Quality of Protection'

configure voip > sbc settings > auth-qop

[AuthQOP]

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.

[0] 0 = The device sends 'qop=auth' in the SIP response, requesting authentication (i.e., validates user by checking user name and password). This option doesn't authenticate the message body (i.e., SDP).
[1] 1 = The device sends 'qop=auth-int' in the SIP response, indicating required authentication and authentication with integrity (e.g., checksum). This option restricts the client to authenticating the entire SIP message, including the body, if present.
[2] 2 = (Default) The device sends 'qop=auth, auth-int' in the SIP response, indicating either authentication or integrity. This enables the client to choose 'auth' or 'auth-int'. If the client chooses 'auth-int', then the body is included in the authentication. If the client chooses 'auth', then the body is not authenticated.
[3] 3 = No 'qop' parameter is offered in the SIP 401 challenge message.

'SBC User Registration Time'

configure voip > sbc settings > sbc-usr-rgstr-time

[SBCUserRegistrationTime]

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.

'SBC Proxy Registration Time'

configure voip > sbc settings > sbc-prxy-rgstr-time

[SBCProxyRegistrationTime]

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.

configure voip > sbc settings > sbc-rand-expire

[SBCRandomizeExpires]

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:

If the received expiry time is less than 610 sec, the device reduces the time by up to 10 sec. For example, if the received expiry time is 110 sec, the device reduces it to anywhere between 100 (i.e., 110 – 10) and 110 sec.
If the received expiry time is greater than 610 sec, the device reduces the time to anywhere between 600 sec and the received expiry time. For example, if the received expiry time is 700 sec, the device reduces it to anywhere between 600 and 700 sec.
Minimum expiry time:
The minimum expiry time that the device can reduce REGISTER messages to is 30 sec and SUBSCRIBE messages to 120 sec. For example, if the received expiry time in a REGISTER message is 35 sec, the device reduces the time to any value between 30 and 35 sec (and not by 10 seconds -- between 25 and 35).
If the received expiry time is less than the minimum (as stated above), the expiry time remains unchanged. For example, if the received expiry time in a REGISTER message is 18 sec, the device forwards the message with this same expiry time (i.e., 18).

Note:

This feature doesn't apply to refresh REGISTER or SUBSCRIBE messages.
You can configure the device to change the received expiry time before forwarding it, using the SBCUserRegistrationTime parameter.

'SBC Survivability Registration Time'

configure voip > sbc settings > sbc-surv-rgstr-time

[SBCSurvivabilityRegistrationTime]

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.

configure voip > sbc settings > sas-notice

[SBCEnableSurvivabilityNotice]

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

[0] = Disable
[1] = Enable

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

'SBC Dialog-Info Interworking'

configure voip > sbc settings > sbc-dialog-info-interwork

[EnableSBCDialogInfoInterworking]

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

[0] Disable (default)
[1] Enable

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

configure voip > sbc settings > sbc-keep-call-id

[SBCKeepOriginalCallId]

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.

configure voip > sbc settings -> sbc-terminate-options

[SBCTerminateOptions]

Defines the handling of in-dialog SIP OPTIONS messages.

[0] = Disabled - the device forwards in-dialog SIP OPTIONS to the outbound peer.
[1] = (Default) Enabled - the device terminates in-dialog SIP OPTIONS and sends a 200 OK response to the peer that sent the OPTIONS message.

'Routing Timeout'

configure voip > sbc settings > sbc-routing-timeout

[SbcRoutingTimeout]

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.

'SBC GRUU Mode'

configure voip > sbc settings > sbc-gruu-mode

[SBCGruuMode]

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

[0] None = No GRUU is supplied to users.
[1] As Proxy = (Default) The device provides same GRUU types as the proxy provided the device’s GRUU clients.
[2] Temporary only = Supply only temporary GRUU to users. (Currently not supported.)
[3] Public only = The device provides only public GRUU to users.
[4] Both = The device provides temporary and public GRUU to users. (Currently not supported.)

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

'BYE Authentication'

configure voip > sbc settings > sbc-bye-auth

[SBCEnableByeAuthentication]

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.

[0] Disable (default)
[1] Enable = The device forwards the SIP authentication response (for the BYE request) to the request sender and waits for the user to authenticate it. The call is disconnected only if the authenticating server responds with a 200 OK.

'SUBSCRIBE Trying'

configure voip > sbc settings > sbc-subs-try

[SBCSendTryingToSubscribe]

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

[0] Disable (default)
[1] Enable

configure voip > sbc settings > sbc-100trying-upon-reinvite

[SBC100TryingUponReinvite]

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

[0] = (Default) Disable
[1] = Enable

configure voip > sbc settings > session-expires-observer-mode

[SipSessionExpiresObserverMode]

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

[0] = (Default) With Grace Time. If the session is not refreshed on time according to the SIP Session-Expires header, the device adds a graceful session expiry time – 120 seconds if the client (UAC) was meant to do the refresh, or 60 seconds if the server (UAS) was meant to do the refresh. If the session is still not refreshed when this graceful time expires, the device disconnects the call (sends a SIP BYE).
[1] = Strict. If the session is not refreshed on time according to RFC 4028 (Session-Expires header time minus the smaller of 32 or 1/3 of the time in Session-Expires header), the call is terminated. For example, if the Session-Expires header is 150 sec., the expected refresh should be done at 150-32 = 118 sec. (as 1/3 of 150 is 50, which is greater than 32). If no refresh is received within this time (e.g., 118 sec.), the device disconnects the call (sends a SIP BYE). Note that the minimum Session Expires is 90 sec.; any received value less than this is set to 90.

'BroadWorks Survivability Feature'

configure voip > sbc settings > sbc-broadworks-survivability

[SBCExtensionsProvisioningMode]

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

[0] Disable = (Default) Normal processing of REGISTER messages.
[1] Enable = Registration method for BroadWorks server. In a failure scenario with BroadWorks, the device acts as a backup SIP proxy server, maintaining call continuity between the enterprise LAN users (subscribers) and between the subscribers and the PSTN (if provided).

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

'SBC Direct Media'

configure voip > sip-interface > sbc-direct-media

[SBCDirectMedia]

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

[0] Disable = (Default) All calls traverse the device (i.e., no direct media).
[1] Enable = Direct media flow between endpoints for all SBC calls.

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

'SBC RTCP Mode'

configure voip > sbc settings > sbc-rtcp-mode

[SBCRTCPMode]

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.

configure voip > sbc settings > sbc-send-invite-to-all-contacts

[SBCSendInviteToAllContacts]

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.

[0] Disable = (Default) Sends the INVITE only to the contact of the received Request-URI.
[1] Enable

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

'SBC Shared Line Registration Mode'

configure voip > sbc settings > sbc-shared-line-reg-mode

[SBCSharedLineRegMode]

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

[0] Disable = (Default) Device forwards the REGISTER messages as is (i.e., not terminated on the device).
[1] Enable = REGISTER messages of secondary lines are terminated on the device.

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

'SBC Forking Handling Mode'

configure voip > sbc settings > sbc-forking-handling-mode

[SBCForkingHandlingMode]

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

[0] Latch On First = (Default) Only the first 18x is forwarded to the INVITE-initiating UA. If SIP 18x with SDP is received, the device opens a voice stream according to the received SDP and disregards any subsequent 18x forking responses (with or without SDP). If the first response is 180 without SDP, the device sends it to the other side.
[1] Sequential = All 18x responses are forwarded, one at a time (sequentially) to the INVITE-initiating UA. If a 18x arrives with an offer only, then only the first offer is forwarded to the INVITE-initiating UA and subsequent 18x responses are discarded.

'Gateway Direct Route Prefix'

configure voip > sbc settings > gw-direct-route-prefix

[GWDirectRoutePrefix]

Defines the prefix destination Request-URI user part that is appended to the original user part for alternative IP-to-IP call routing from SBC to Gateway (Tel) interfaces.

The valid value is a string of up to 16 characters. The default is "acgateway-<original prefix destination number>". For example, "acgateway-200".

For more information, see Configuring SBC IP-to-IP Routing Rules.

configure voip > sbc settings > sbc-media-sync

[EnableSBCMediaSync]

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).

[0] = (Default) Disable. Media synchronization is performed only if the RTP mode (e.g., a=sendrecv, a=sendrecv, a=sendonly, a=recvonly, and a=inactive) between the user agents are different and synchronization is required.
[1] = Enable. Media synchronization is performed if the media, including RTP mode or any other media such as coders, is different and has not been negotiated between the user agents.
[2] = Never. Media synchronization is never performed.

'Remove SIPS from Non-Secured Transport'

configure voip > sbc settings > sbc-remove-sips-non-sec-transp

[SBCRemoveSIPSFromNonSecuredTransport]

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.)

[0] Disable = (Default) The device replaces “sips:” with “sip:” for the Request-URI and Contact headers only (and retains “sips:” for all other headers).
[1] Enable = The device replaces “sips:” with “sip:” for the Request-URI, Contact, From, To, P-Asserted, P-Preferred, and Route headers.

'SIP Topology Hiding Mode'

configure voip > sbc settings > sip-topology-hiding-mode

[SIPTopologyHidingMode]

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:

Headers concerned with the source of the message are overwritten with the IP address of the IP Interface from where the device sends the message.
Headers concerned with the destination of the message are overwritten with the destination IP address.

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

[0] By Host Name Parameters Only = (Default) The device overwrites the host part in the SIP headers according to the configuration of the IP Group's 'SIP Group Name' and 'SIP Source Host Name' parameters. If these parameters are empty, the device doesn't overwrite the host part of the headers.
[1] Fallback to IP Addresses = This option is applicable only to dialog-initiating requests and in-dialog REFER requests.
If the 'SIP Group Name' parameter of the destination IP Group is empty, the device overwrites the host part of the following destination-related SIP headers with the destination IP address: Request-URI and P-Called-Party-ID for all types of requests, and To header for non-REGISTER requests. If the 'SIP Group Name' parameter is configured, the device overwrites the host part in these headers with the configured value.
The source-related headers which are overwritten when the 'SIP Source Host Name' parameter is configured (From, P-Asserted-Identity, P-Preferred-Identity, Referred-By, P-Charge-Info, Remote-Party-ID, P-Associated-URI, Diversion, and History-info) are always overwritten. If the 'SIP Source Host Name' parameter of the destination IP Group is configured, the device overwrites the host part with the configured value. If the 'SIP Source Host Name' parameter of the destination IP Group is empty, the device overwrites the host part of the mentioned headers with the IP address of the device's IP Interface from where it sends the message.

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

'Enable MSRP'

configure voip > sbc settings > enable-msrp

[EnableMSRP]

Enables Message Session Relay Protocol (MSRP).

[0] Disable (default)
[1] Enable

For more information, see Configuring Message Session Relay Protocol.

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

Push Notification Service

configure voip > sbc settings > pns-reminder-period

[PNSReminderPeriod]

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.

configure voip > sbc settings > pns-register-timeout

[PNSRegisterTimeout]

Defines the maximum time (in seconds) that the device waits for a SIP REGISTER refresh message from the user, before it forwards an incoming SIP dialog-initiating request (e.g., INVITE) to the user.

The valid value range is 5 to 50. The default is 30.

When the device receives an incoming SIP dialog-initiating request whose destination is the user, it sends an HTTP message to the Push Notification Server to trigger it into sending the user a push notification so that the user can send a REGISTER refresh message to the device. If the device receives the REGISTER refresh message within this timeout, it forwards the incoming SIP request to the user. If the timeout expires and the device still hasn't received the REGISTER refresh message, the device rejects the call.