In this section:
Not applicable to
Creating a SIP Signaling Group
Create a SIP Signaling Group:
From the Create Signaling Group drop down box, select SIP Signaling Group.
Modifying a SIP Signaling Group
- Click the expand ( ) Icon next to the entry you wish to modify.
- Edit the entry properties as required, see details below.
Configuring a SIP Signaling Group
Click '
' on the WebUI screen to configure additional itemsCreate the following configuration resources used for the SIP Signaling Group directly from the SIP Signaling Group screen, by clicking the '
' icon for the field in the WebUI- Call Routing Table
- Local Registrar Table (if you select Forward Registration, Forward Registration after Local Processing, or Local Registrar for the SIP Mode)
- Media List ID
- SIP Server Table (if you select Basic Call, Forward Registration, or Forward Registration after Local Processing for the SIP Mode)
- Tone Table
Field definitions for the Create SIP Signaling Group window are listed by feature.
Quick Links - Configuration Options by Feature
Configure a SIP Signaling Group:
For
Unable to show "metadata-from": No such page "_space_variables"only: from the Create Signaling Group drop-down box, select SIP Signaling Group.For the
Unable to show "metadata-from": No such page "_space_variables"only: click Add SIP SG:Configure the field options. Field definitions below are listed in SIP Signaling Group feature groups. See Quick Links - Configuration Options by Feature.
Overview - Field Definitions
Field | Description |
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Description | Descriptive name for the signaling group. |
Admin State | Specifies the admin state of the Signaling group.
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SIP Channels and Routing - Field Definitions
Field | Description | |||||||||||||||
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Action Set Table | Specifies a defined Action Set Table for this Signaling Group. | |||||||||||||||
Call Routing Table | Specifies the Call Routing Table this Signaling Group uses. | |||||||||||||||
Number of Channels | Specifies the number of SIP channels available for calls in this Signaling Group. Valid entry: 1 - 960. | |||||||||||||||
SIP Profile | Specifies the SIP Profile to use for this Signaling Group. | |||||||||||||||
SIP Mode | Specifies the SIP Registration Mode to use in this Signaling Group.
If Fwd. Reg. After Local Processing is configured as the SIP Mode , the selected SIP Server Table for that same Signaling Group should not be configured with a Contact Registrant Table. The SBC does not support Fwd Reg. After Local Processing and a Contact Registrant Table in the same Signaling Group. See also: Configuring the SBC Edge Portfolio for Site Survivability. | |||||||||||||||
SIP Mode (SBC SWe Edge only) | Specifies the SIP Registration Mode to use in this Signaling Group.
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Agent Type | Specifies the way in which SIP-SIP signaling will be handled by the SBC. Valid entries: Access Mode: When Access Mode is enabled, the SBC allows certain SIP functions to be sent in passthrough mode:
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. Back-to-Back User Agent. The SBC will maintain state and participate in all SIP signaling between both endpoints. Default selection: Back-to-Back User Agent. Access Mode SG Requirement
If Access Mode is specified, a source and destination signaling group must exist (e.g., Broadsoft Local Register SG to Broadworks Server SG). | |||||||||||||||
Interop Mode | 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: 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. Applicable to SBC 1000/2000 only. 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. Applicable to SBC 1000/2000 only. Standard. SBC will interoperate according to RFC standards. Default entry: Standard. When the Agent Type is set to Back-to-Back User Agent, the BroadSoft Extension option is not available. | |||||||||||||||
Office 365 User Domain Suffix | Note
This field is applicable to SBC 1000/2000 only. 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. | |||||||||||||||
Registrant TTL | Specifies the Time-To-Live (TTL) value for inbound registration. Inbound registration values should be equal to or greater than this. Valid entry: Enter a value in seconds. Default value: 3600. This field is visible when the Agent Type is set as Access Mode. | |||||||||||||||
AD Attribute | Note
This field is applicable to SBC 1000/2000 only. 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. | |||||||||||||||
AD Update Frequency | Note
This field is applicable to SBC 1000/2000 only. 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. Note
As an option, the AD Live Update feature forces the SBC to query AD directly to obtain updated records (rather than following the parameter set in the AD Update Frequency). | |||||||||||||||
AD First Update Time | Note
This field is applicable to SBC 1000/2000 only. 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. | |||||||||||||||
Registrar | Specifies the Registrar Table attached to the Signaling Group for routing purpose and adding registration records. | |||||||||||||||
Registrar Min. TTL | 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. | |||||||||||||||
Outbound Registrar TTL | Specifies the Time-To-Live (TTL) value for outbound registrations. This is configured only in the Forward modes of operation only. | |||||||||||||||
SIP Server Table | 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. | |||||||||||||||
Load Balancing | Specifies the load balancing method used for SIP Server registration and redundancy in the Signaling Group.
Note
For the Failover SIP Registration feature, the following options must be enabled:
The type of the SIP Server: IP/FQDN or DNS-SRV determines the available options for Load Balancing:
Note
In the SIP Server configuration, the Priority configuration option determines the order in which a SIP Server is used for redundancy and registration. Refer to Creating and Modifying Entries in SIP Server Tables. | |||||||||||||||
Channel Hunting | Specifies the method that Call Control uses to allocate SIP channels.
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Notify Lync CAC Profile | Enables whether any CAC Profile updates received locally are transmitted to the SIP servers listed in this Signaling Group Configuration. Valid entry: Disable (Default, updates received locally are not transmitted to SIP servers) or Enable (updates received locally are transmitted to SIP servers). | |||||||||||||||
Challenge Request | 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.
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Outbound Proxy | Specifies the outbound proxy through which all SIP messages are sent. For in-depth configuration detail, see Outbound Proxy Configuration. | |||||||||||||||
Outbound Proxy Port | Specifies the port number for the outbound proxy, if one is configured. The port number must be in the range 1024 through 65535. | |||||||||||||||
No Channel Available Override | Note
This field is applicable to SBC 1000/2000 only. In the event of a call release due to the No Channel/Circuit available release cause code, the specified cause code is sent to the relevant protocol module. For more information, refer to the list of Cause Codes. Note
This attribute should be applied to the inbound Signaling Group of calls. Even in the case of an outbound Signaling Group having no channel available, the parameter value on the inbound Signaling Group is used to determine the cause code to send. See below for an example of this behavior: | |||||||||||||||
Call Setup Response Timer | Specifies the interval of time, in seconds, after a call is initiated that the Unable to show "metadata-from": No such page "_space_variables" (SBC) waits for a call to connect before terminating the incoming call. | |||||||||||||||
Call Proceeding Timer | 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. | |||||||||||||||
QoE Reporting | Note
This field is applicable to SBC 1000/2000 only. 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 | |||||||||||||||
Use Register as Keep Alive | Use Registration requests as Keep Alive for marking the Signaling group as up or down. Default entry: Enabled. 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. | |||||||||||||||
Forked Call Answered Too Soon | When enabled, the Unable to show "metadata-from": No such page "_space_variables" ignore a forked leg (disconnect) if that leg is going to a voicemail box, but the rest of the forked legs may connect to the subscriber's other numbers. For details, refer to Forked Calls Answered To Soon (Disconnect on Quick Connect).Possible entries:
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Answer Too Soon Timer | This field is available only when Forked Call Answered Too Soon is Enabled. Specifies the timer for the forked leg to be disconnected. Valid entry: 1 - 5000 ms. | |||||||||||||||
SIP Recording Status | For SIPREC, enables or disables the SIP Recording functionality on the selected signaling group. Select from the drop down list.
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SIP Recorder | When SIPREC is enabled for the specific Signaling Group, this field enables the selection of an entry from a SIP Recording Table. Select from the drop down list. This list is populated from the entries configured in the SIP Recording table. For details, refer to Creating and Modifying Entries in the SIP Recorder Table. This field is available only when SIP Recording Status is set to Enabled. |
Media Information - Field Definitions
Field | Description |
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Audio/Fax Stream Mode | Note
This field available is for SBC 1000/2000 only. For Unable to show "metadata-from": No such page "_space_variables" , see Supported Audio Modes .Determines the streaming mode for audio, fax, and media transmission.
All options are enabled by default. |
Supported Audio Modes | Note
This field available is for SBC SWe only. For SBC 1000/2000 see Audio/Fax Stream Mode. Determines the streaming mode for audio, fax, and media transmission.
All options are enabled by default. |
Proxy Local SRTP Crypto Profile ID | Note
Available for Unable to show "metadata-from": No such page "_space_variables" only. This field is available only when Proxy with Local SRTP is included in the Supported Audio mode list. Determines whether a call crypto file is used with SRTP media negotiation.
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Video/Application Stream Mode | Determines the streaming mode for audio, fax, and media transmission. This feature requires a video license.
The first two options are enabled by default. |
Media List ID | Specifies the Media List used by this Signaling Group. |
Play Ringback | Specifies how ringback plays on a channel. Note
The Play Ringback configuration functions with 180 Ringing or ISDN Alert. The function also executes when SIP 183 is received with Session Description Protocol (SDP).
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Tone Table | Specifies the Tone Table used by this Signaling Group. Only visible if Always or Auto is specified for Play Ring Back. |
Play Congestion Tone | Specifies whether a congestion tone plays when a 503 response with reason header Q.850 and cause code = 42 is received for outbound INVITE. Options: Enable (congestion tone is plays) or Disable (default; congestion tone does not play). |
Early 183 | 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 (e.g. Mediation server) from staying in the Trying state. This setting is required for Skype for Business interoperability. Early 183 is applicable only when Audio/Fax Stream DSP Mode is enabled as the media mode. |
Allow Refresh SDP | Specifies whether to allow refresh SDP after the media has been negotiated. Options: Enable (enables SDP after negotiation) or Disable (does not allow SDP after negotiation). Default entry: Enable. |
Music on Hold | The field enables Music on Hold at the SIP Signaling group level. Available options:
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 Portfolio. |
RTCP Multiplexing | Note
For details on how to use Real-Time Transport Control Protocol (RTCP) Multiplexing with Microsoft Teams Direct Routing, refer to the following pages: Enables RTCP Multiplexing to combine two parts of the Real-Time Transport Protocol (RTP) for data traffic and RTCP (for control information) onto a single multiplexed UDP port. RTCP Multiplexing is supported according to RFC 5761, in which the SDP contains the value of a=rtcp-mux. RTCP Multiplexing is supported for DSP mode only. Valid entries: Enable ( SDP contains an a=rtcp-mux attribute, and RTP and RTCP are able to be combined on a single multiplexed port if the far end is capable ) or Disable ( SDP does not contain an a=rtcp-mux attribute and RTP and RTCP consume two ports). Default value: Disable. The peer endpoint must support the a=rtcp-mux exchange for the RTP and RTCP port multiplexing onto one data port. |
Media Codec Latch | Media Codec Latch allows enabling or disabling of the codec lockdown when receiving multiple codecs in offer answer model. When disabling Media Codec Latch, the SBC will not renegotiate the codec if the peer sends more than one codec in the answer. Valid entries: Enable or Disable Default value: Enable |
Mapping Tables- Field Definitions
Field | Description |
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SIP To Q.850 Override Table | Specifies the SIP to Q.850 Override Table to use for this Signaling Group. |
Q.850 To SIP Override Table | Specifies the Q.850 To SIP Override Table to use for this Signaling Group. |
Pass-thru Peer SIP Response Code | 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. |
SIP IP Details - Field Definitions
Field | Description |
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Teams Local Media Optimization | Microsoft Teams Direct Routing media flow uses the shortest path to improve the SBC Edge's media quality and bandwidth usage. |
Signaling/Media Source IP | Specifies the Logical IP address at which SIP messages are received. This address is used as the source IP for all SIP messages leaving the Unable to show "metadata-from": No such page "_space_variables" or SBC 1000/2000 through this Signaling Group. The physical interface on which these messages leave the system is determined by the System IP routing configuration. The IP version (IPv4 or IPv6) for the SIP Server used by this Signaling Group determines the IP version used for all outgoing messages leaving the SBC through this Signaling Group.Note
If Static NAT is used, the configured NAT Public IP replaces the Source IP selected or acquired when set to Auto mode. When when the Outbound NAT Traversal field is set to Static NAT, the name of the Signaling/Media Source IP option changes to Signaling /Media Private IP.
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Private Media Source IP | Private IP address that handles routing of messages for media. This field is displayed only when Teams Local Media Optimization is enabled. Valid entry: select an IP address from the drop down list. |
Signaling DSCP | Each SIP-SG is configurable with the DSCP value to be used for signaling. This allows for improved quality of service in real-time applications, such as conferencing and conversations. The settings take effect for both client and server modes of SIP. The default value of 40 is the most common value used in the VOIP networks for signaling packets. The configured value should be chosen according to the QoS policies of the IP network in which the signaling packets travel. Valid entry range: 0 to 63 (inclusive). Default value: 40. |
NAT Traversal | Specifies whether or not the Signaling Group uses a third-party entity IP address inside SIP message to support network address translation (NAT). Only visible when NAT Traversal is set to Static NAT.
Symmetric NAT (port forwarding) is the only supported NAT type. This NAT configuration type means that packets received on a specific NAT server port are always forwarded to the same SBC port, for example, packets on the NAT public IP, port 5060 are forwarded to a private (SBC) IP, port 5060. |
ICE Support | Specifies whether ICE support is enabled/disabled. Enable/disable is displayed only when SIP Mode is configured to Basic Call.
When Interactive Connection Establishment (ICE) support is enabled, it takes precedence over all other media related NAT configuration. |
ICE Mode | Specifies the Interactive Connection Establishment (ICE) Mode that is enabled/disabled. This field is displayed only when ICE Support is enabled. For SBC 1000/2000: Two ICE Modes are available:
For Unable to show "metadata-from": No such page "_space_variables" :
For details on ICE, refer to Working with the SBC Edge Portfolio and SBC SWe Edge as an ICE-Lite Agent and Working with the SBC Edge Portfolio as an ICE Agent. |
Static NAT - Outbound | If the selected Outbound NAT Traversal is Static NAT, you must enter the field NAT Public IP (Signaling/Media) appears. |
Static NAT - Inbound | |
Detection | Enables and disables NAT Traversal detection for inbound SIP/RTP packets.
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Qualified Prefixes Table | 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. |
Secure Media Latching | 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. |
Source Media Netmask | Specifies netmask used to compare the SIP Signaling IP and the RTP IP used for latching.
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Registrar Max. TTL Enabled | 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. |
Registrar Max. TTL | 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.
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Application Layer IP | Specifies how the Signaling Group will select the local IP.
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NAT Public IP Address | Specifies the public IP of the NAT server visible from Internet. The NAT server's ports must be configured to allow SIP and RTP traffic, for example: port range 5060-5061 for SIP and 16000-17000 for RTP. The IP address specified in this field must be publicly accessible. |
Listen Ports - Field Definitions
This section defines a listening port and protocol for the SIP Signaling Group.
Field | Description |
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Listen Port | This section defines a listening port and protocol for the SIP Signaling Group. The Listen Ports populated in this screen are derived from the configuration in the Listen Port Table. The Signaling Group uses this port for communication with external peers.
Note
For the SfB to Microsoft Teams migration, do not configure the Listen Ports table of the SfB signaling group with the same Listen Ports of the Microsoft Teams signaling group. |
Federated IP/FQDN - Field Definitions
Federated IP addresses and FQDNs specified in a SIP Signaling Group are permitted.
The Federated IP/FQDN feature acts as an access control by defining from which server a SIP Signaling Group will accept messages.
Field | Description |
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IPv4/6 Address or FQDN | 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:
For IPv6:
For FQDN:
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Federated IP Netmask/Prefix | For IPv4: Specifies the network address mask to apply against the specified server address. For IPv6: Specifies the prefix to identify the subnet. |
Message Manipulation - Field Definitions
Field | Description |
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Message Manipulation | 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). |
Inbound Message Manipulation | 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).
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Outbound Message Manipulation | 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).
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