Creating and Modifying SIP Signaling Groups

In this section:

Specifies the admin state of the Signaling group.

• Enable: Enables the signaling group.
• Disable: Disables the signaling group.
• Drain: When Drain is selected, calls that are currently up remain connected. However, new calls and forking are not allowed for the Signaling Group. This allows all the calls to be drained from the SG.

Specifies a defined Action Set Table for this Signaling Group.

Specifies the Call Routing Table to be used by this Signaling Group.

Specifies the number of SIP channels available for calls in this Signaling Group.

For SBC 1000/2000. Valid entry: 1 - 960.

For SBC SWe Lite. Valid entry: 1 - 1000.

Specifies the SIP Profile to use for this Signaling Group

Specifies the SIP Registration Mode to use in this Signaling Group.

• Basic Call: Sends INVITEs to the selected server table.
• Forward Reg. After Local Processing: Forwards the REGISTER request after inserting it to the local registrar.
• Local Registrar: Maintains local registrar only. Uses registrar bindings to terminate a call.

See also: Configuring the SBC Edge for Site Survivability.

Specifies the way in which SIP-SIP signaling will be handled by the SBC. Valid entries:

Back-to-Back User Agent. The SBC will maintain state and participate in all SIP signaling between both endpoints. 

Access Mode:  When AccessMode is enabled, the SBC allows certain SIP functions to be sent in a pass through mode:

• Non-call related methods (Register, Subscribe and Notify)
• Refer and Update
• Authentication handling

When Agent Mode is set to Access Mode, the Signaling Group service status will always show as "green" or "up" to allow for pass through messaging. Also, when Agent Type is set to Access Mode, the options Office 365 and Office 365 with AD PBX are disabled.

Default selection: Back-to-Back User Agent.

Indicates if the SBC should interoperate in a proprietary manner for certain functionality. Multiple SIP Signaling Groups can enable this mode, but this mode can only be enabled once for each SIP Signaling Group/SIP Server combination.

Valid entries:

Standard. SBC will interoperate according to RFC standards.

BroadSoft Extension. SBC will support BroadSoft related extensions. Specifically, the SBC will use BroadSoft’s subscriber data (BroadWorksSubscriberData) while in remote survivability mode. This feature allows the SBC to retrieve and store alternative user information for use when the BroadSoft server is unreachable.

Office365. Indicates the Signaling Group is communicating directly with the Skype for Business Front End Pool.

Office 365 w/AD PBX. Indicates the Signaling Group is communicating directly with the Skype for Business Front End Pool. The SBC retrieves AD records (based on the entry in the AD Attribute field), and uses that to register on-premises PBX endpoints to Skype for Business Front End Pool. 

Header Transparency (available in SWe Lite only). Indicates the SBC passes through all non-dialog related SIP headers from any inbound SIP messages to the outbound messages.(If this field is configured as Broadsoft Extension, the SBC passes through only the SIP headers required by Broadsoft from inbound SIP messages to outbound SIP messages).

Default entry: Standard.

The FQDN used in the Signaling Group to communicate directly with the Skype for Business Front End Pool.

This field is visible when the Interop Mode is set to Office 365 or Office 365 w/AD PBX.

This field can be NULL; if so, the address of the SIP Server will be used.

Specifies the Time-To-Live (TTL) value for inbound registration. Inbound registration values should be equal to or greater than this. Valid entry: Enter value in seconds. Default value: 3600.

This field is visible when the Agent Type is set as Access Mode.

Specifies any desired Active Directory attribute name in which the PBX number to be Registered is located; this field is dependent on how AD is configured. Default entry: =pager=.

This field is visible when Interop Mode is set as Office 365 w/AD PBX.

Controls the frequency (in days) for how often SIP queries Active Directory for all records with the specific AD attribute populated. Valid entry: 1 - 30 days.

This field is visible when Interop Mode is set as Office 365 w/AD PBX.

Specifies time of first AD query update in hh:mm:ss (24-hour format). Valid entry: hh (hour), mm (minute), ss (seconds).

This field is visible when Interop Mode is set as Office 365 w/AD PBX.

Specifies the Registrar Table attached to the Signaling Group for routing purpose and adding registration records.

Specifies the Time-To-Live (TTL) value for inbound registrations. Inbound registration values should be equal to or greater than this. This is configured in all registration scenarios.

Specifies the Time-To-Live (TTL) value for outbound registrations. This is configured only in the Forward modes of operation only.

Specifies the SIP Server Table to be used in the Signaling Group. The options in this field are derived from the configuration of SIP Server tables, see Creating and Modifying Entries in SIP Server Tables.

If a SIP Server Table is added which includes a server that has Stagger Registration enabled, Stagger Registration occurs. Also, if a SIP server table is removed which included a server that had Stagger Registration enabled, Stagger Un-registration occurs. For more information about Stagger Registration, see Creating and Modifying Entries in SIP Server Tables .

If you select an entry in the SIP Server table that is defined as DNS SRV, the Load Balancing field is not visible. See Load Balancing.

This field applies only when the SIP Server Table selected from the drop down list is IP/FQDN based and does not contain SRV servers (load balancing is not required if the SIP Server Table which contains a SRV server). See SIP Server Table.

Specifies the load balancing method used with this Signaling Group. Used only in the SIP Basic Call Mode.

• Round Robin: Each initial INVITE sent to the next server in the pool. As the basic algorithm, the scheduler selects a resource pointed to by a counter from a list, after which the counter is incremented and if the end is reached, returned to the beginning of the list. Round-robin selection has a positive characteristic of preventing starvation, as every resource will be eventually chosen by the scheduler.
• Priority: The request goes to the server with highest priority.
• First: The initial request goes to the first available server.

Specifies the method that Call Control uses to allocate SIP channels.

• Standard: Specifies the first available low numbered channel.
• Reverse Standard: Specifies last available high numbered channel.
• Round Robin: Specifies channels based on next available from low numbered to high numbered.
• Least Idle: Specifies that channels are chosen based on the least idle channel.
• Most Idle: Specifies that channels are chosen based on the most idle channel.

Enables whether any CAC Profile updates received locally are transmitted to the SIP servers listed in this Signaling Group Configuration. Valid entry: Enable (Default, updates received locally are transmitted to SIP servers), or Disable (updates received locally are not transmitted to SIP servers) .

Indicates whether or not incoming request messages are challenged for security purposes. If this option is set to Enable, you must specify a realm and at least one entry in the Authorization Table.

• True: All requests are challenged for realm, user, and password.
• False: No request messages are challenged.

Specifies the outbound proxy through which all SIP messages are sent. For in-depth configuration detail, see Outbound Proxy Configuration.

Specifies the port number for the outbound proxy, if one is configured. The port number must be in the range 1024 through 65535.

In the event of a No Channel/Circuit available release cause code, the specified cause code is sent to the relevant protocol module. For more information see the list of Cause Codes.

NOTE: This attribute relates only to Inbound calls on the Signaling Group where applied.

Specifies the interval of time, in seconds, after a call is initiated that the SBC Edge (SBC) waits for a call to connect before terminating the incoming call.

Timer indicates the amount of time to wait after receiving a “100 Trying” for a call attempt (egress INVITE). When the timer expires the call will not proceed.

Option: 24 - 750 seconds.

Enables the QoE (Quality of Experience) reporting feature in the SBC. Valid options: Enabled (enables the feature) or Disabled (disables the feature). This field must be enabled for the QoE options to be available through the QoE Settings. See Configuring Quality of Experience (QoE) Settings

When this field is set to Disabled, only SIP Options (if configured) are used as a keep alive mechanism to mark the Signaling group as up or down

Determines the streaming mode for audio, fax, and media transmission.

• DSP: Audio/fax stream is processed using a DSP resource, and consumes a DSP license.
• Proxy: Audio/fax stream is processed using the SBC internal processing resources and does not require a DSP resource or license.
• Direct: Audio/fax stream is passed directly from one compatible end-point to another, this uses neither the SBC processor or a DSP resource.

All options are enabled by default.

Determines the streaming mode for audio, fax, and media transmission.

• DSP: Audio stream is processed using a DSP resource, and consumes a DSP license.
• Proxy Relay: Audio stream is processed using the SBC internal processing resources and does not require a DSP resource or license; SRTP is untouched locally by the SBC.
• Proxy with Local SRTP. Media stream is switched between endpoints using SRTP media encryption on a call leg basis. Note: The SBC SWe Lite supports 1,000 RTP Proxy Relay signaling session with no additional licensing. Support of Proxy with Local SRTP (encryption/decryption) requires licensing up to 1,000 signaling sessions  See Working with Licenses.
• Direct: Audio stream is passed directly from one compatible end-point to another, this uses neither the SBC processor nor a DSP resource.

All options are enabled by default.

Determines whether a call a crypto file is used with SRTP media negotiation.

Crypto File. Indicates the crypto file profile used to negotiate SRTP.

None. SRTP is not used for media negotiation.

Determines the streaming mode for audio, fax, and media transmission. This feature requires video license.

• Proxy: the video/application stream is processed using the SBC internal processing resources.
• Direct: the video/application stream is passed directly from one compatible end-point to another, this uses neither the SBC processor.
• Disabled: the route is not used for video calls.

The first two options are enabled by default.

Specifies the Media List used by this Signaling Group.

Specifies how ringback is played on a channel.

• Auto on 180 (applies to 180/Alert only) or Auto on 180/183 (applies to 180/Alert and 183/Progress): Configured for Auto, the SBC operates in the nominal RFC 3960 pattern
  • SBC generates ringback until in-band media arrives.
  • SBC-generated ringback is discontinued in the presence of in-band ringback.

  • Additionally, any ALERT is sent with PI=8 regardless of whether or not a SDP was received on the SIP side. Doing so allows the SBC to send in-band audio without signaling PROGRESS.

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• Always on 180 (applies to 180/Alert only) or Always on 180/183 (applies to 180/Alert and 183/Progress): SBC provides ringback, ignoring any arriving in-band media.
  • For SIP-originated call legs, 180w/SDP will be sent to permit SBC to provide in-band ringback via early media.

  • For ISDN-originated call legs, ALERT+PI will be sent along with SBC-inband ringing.

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• Never: SBC does not provide ringback and cuts through ringback from the source when/if it arrives.

  • With NEVER, SBC will send 180/ALERT to the originating call leg without SDP/PI as in this configuration SBC will not supply inband ringback.

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Note: The Play Ring Back setting is activated only after the channel receives an ALERT or 180 Ringing. Issues with ringback and 183 Session Progress must be addressed using a Message Translation.
Click to see more information about Activating Play Ring Back.

Specifies the Tone Table used by this Signaling Group. Only visible if Always or Auto is specified for Play Ring Back.

Specifies whether a congestion tone is played when a 503 response with reason header Q.850 and cause code = 42 is received for outbound INVITE.

Options: Enable (congestion tone is played) or Disable (default; congestion tone is not played).

Specifies whether to send a SIP 183 response immediately after receiving an Invite message. The early 183 Session Progress with SDP provides the SRTP key that will be used to decrypt the transmit stream from SBC to the SIP peer. This setting is used to prevent the peer device (eg. Mediation server) from staying in the Trying state. This setting is required for Lync 2010/2013 Skype for Business interoperability.

Early 183 is applicable only when Audio/Fax Stream DSP Mode is enabled as the media mode.

The field enables Music on Hold at the SIP Signaling group level. Available options:

• Always Enabled: Enables local Music on Hold (for SDP attribute “sendonly” or “inactive”).
• Enabled for SDP Inactive: Enables local Music on Hold when there is no possibility of the peer device sending its own music during hold (for SDP attribute “inactive”).
• Disabled: Disables Music on Hold for the SIP Signaling Group.

Default value: Disabled.

For detailed information about enabling Music On Hold as part of the Media configuration, see Configuring the Media System. For detailed information about uploading music files, see Uploading Music on Hold Files - SBC Edge.

Specifies the SIP to Q.850 Override Table to be used for this Signaling Group.

Specifies the Q.850 To SIP Override Table to be used for this Signaling Group.

The default value is Enabled. If you disable the pass-thru peer SIP response, then the mapping tables will be applied to SIP-SIP calls.

Specifies the the Logical Port IP address or VLAN sub-interface* over which SIP session are conducted. Only visible when NAT Traversal is set to None.

• Auto: The node automatically selects the port IP address or VLAN sub-interface* over which the session is conducted; either IPv4 or IPv6 address is sent (based on the SIP Server's IP address).
• Ethernet IP : Specifies the logical Ethernet port IP address or VLAN sub-interface* over which the session is conducted. Both IPv4 and IPv6 addresses are supported.

*Note:  VLAN sub-interfaces are listed in the Signaling/Media Source IP drown down list for SWe Lite only. For VLANs on the SBC 1000 or SBC 2000, see Managing VLANs.

Enables and disables NAT Traversal detection for inbound SIP/RTP packets.

• When detection is disabled, none of the following fields are visible.

Specifies which SIP NAT Qualified Prefix Table to use in association with this SIP Signaling Group. The Qualified Prefixes Table is used to determine whether or not a particular subnet is behind a NAT device. If None is selected from the Qualified Prefixes Table drop down list, then all subnets are treated as if they were behind a NAT device. The options available from this drop down list are configured as part of NAT Qualified Prefixes. 

Enables and disables Secure Media Latching for inbound RTP packets. When enabled, media latching occurs only if the RTP packet's IP is in the same subnet as the public IP seen by SIP signaling.  When disabled, no IP address security checks are performed during RTP latching.

Specifies netmask used to compare the SIP Signaling IP and the RTP IP used for latching.

• The Source Media Netmask field is present if Inbound NAT Transversal Detection and Secure Media Latching are enables.
• The netmask must be specified in dotted decimal format (e.g., 255.255.255.255).

Enables and disables the time to live (TTL) functionality for inbound registrants from behind a NAT.  If a client registers with an expires value greater than the value specified in the Registry Max. TTL field, the expiration is adjusted to the value specified in the Registrar Max TTL field.

Specifies a maximum time to live (TTL) for the SIP registration. The SBC uses this feature to determine that the client is still active, and aid in keeping bindings to remote NAT devices alive. If the SBC does not receive a request to re-register from the client before the expiry, the call and the registration are torn down.

• Note that this field is only applicable to clients registering from behind NAT devices.
• Setting the Max. TTL to a low duration may result in excessively high system resource usage as the number of NAT clients increases.

Specifies how the Signaling Group will select the local IP.

• Auto: Means SBC connects to the peer and queries the local IP which is used in SIP headers and SDP.
• Bind: Binds to specified interface and uses that IP (1st/2nd) in headers and SDP.

Specifies the port to listen for SIP messages.

Specifies the protocol with which this port can receive SIP messages.

If TLS is selected this specifies the TLS Profile this port will use for secure SIP messages.

Specifies the IP Address (IPv4 or IPv6) or Fully Qualified Domain Name of a server from which the SBC will accept SIP messages. Federated IP allows IPv4, IPv6, or FQDN address format.

For IPv4:

• If the Netmask is not specified, 255.255.255.255 is used.
• When an IP Address is specified the Netmask is mandatory.

For IPv6:

• Prefix format is used to identify the subnet.

For FQDN:

• When an FQDN is specified, all the IPs in that domain are added.
• Uses the Federated IP Netmask/Prefix field to determine whether the signaling group uses IPv4 or IPv6.

For IPv4: Specifies the network address mask to apply against the specified server address.

For IPv6: Specifies the prefix to identify the subnet.

This option enables or disables the ability for the SBC to manipulate SIP messages using previously configured Message Tables.

Select from the drop down list: Enable (enables the feature) or Disable (disables the feature).

The rules in this table are used to manipulate inbound SIP messages in the Signaling Group. The Signaling Group will support a maximum of 10 Message Rule Tools allowed in the Signaling Group (inbound direction).

Up. Moves the message table entry up in the list.The rules are applied in the order the tables are listed.

Down. Moves the message table entry down in the list.The rules are applied in the order the tables are listed.

Add. Displays a drop down list of available message tables. Select an entry and click Apply.

Remove. Removes the message table entry from the list.

The rules in this table are used to manipulate outbound SIP messages in the Signaling Group. The Signaling Group will support a maximum of 10 Message Rule Tools allowed in the Signaling Group (outbound direction).

Up. Moves the message table entry up in the list. The rules are applied in the order the tables are listed.

Down. Moves the message table entry down in the list.The rules are applied in the order the tables are listed.

Add. Displays a drop down list of available Message Tables. Select an entry and click Apply.

Remove. Removes the message table entry from the list.