To create or modify an Entry to a Call Routing Table:
- In the WebUI, click the Settings tab.
In the left navigation pane, go to Call Routing > Call Routing Table > Entry.
Modifying an Entry to a Call Routing Table
- Click the expand (
) Icon next to the entry you wish to modify. - Edit the entry properties as required, see details below.
Resequencing an Entry in a Call Routing Table
- Click the Resequence icon (
) at the top of the table. - Select the row(s) you want to move.
- Click the Move Selected Rows Up (
) or Move Selected Rows Down ( 
) icon to reposition the row(s) in the table. - Click Apply.
Enabling an Entry in a Call Routing Table
- Select the check box next to the entry you wish to enable.
- Click the Enable (
) icon at the top of the table.
Disabling an Entry in a Call Routing Table
- Select the check box next to the entry you wish to disable.
- Click the Disable (
) icon at the top of the table.
Creating an Entry to a Call Routing Table
Click the Create Routing Entry (
) icon.
Call Routing Entry - Field Definitions
Click '
' on the WebUI screen to configure additional items
Configuring an Object within an Object
Access a WebUI page that includes a field for a configuration resource with support for object within an object configuration. For example, the SIP Signaling Group Table includes a number of such fields.
Click the field-specific (
) icon next to a specific object. In the example below, the Call Routing Table field supports object within object configurability:
At the Create window, enter a description for the newly created object.
Click OK.
The Call Route Table is created. The field is auto-populated with the newly-created table, and the "changed field" is highlighted (i.e., the Call Routing Table field).
| Field Definition | Description |
|---|---|
Admin State | Specifies the admin state of the Call Route. Valid entry: Enable (enables the call route entry for routing the call, displays in configuration header as |
Route Priority | Priority of the route from 1 (highest) to 10 (lowest). Higher priority routes are matched against before lower priority routes regardless of the order of the routes in the table. |
Call Priority | Call Priority is used for emergency (911) calls in the U.S. and other countries. This field is used if the Priority header is not received in the ingress SIP INVITE and for PSTN interfaces. The default value is Normal. When integrating with E911 providers, Priority may have to be set to Emergency. |
Number/Name Transformation Table | Specifies the Transformation Table to be used for this routing entry. This drop-down list is populated from the entries in the Transformation Table. |
| Time of Day Restriction | Select an optional time of day table to use with the call route. All Time of Day tables are included in this drop-down list. For details on Time of Day, refer to Working with Time of Day. |
| Destination Type | Specifies the destination type for calls using this route. Valid selections: Standard, Registrar Table, Deny, or Trunk Group. Standard. Call routes to normal types such as ISDN or SIP signaling groups. Specify a list of signaling groups through Destination Signaling Groups. Registrar Table. Call routes to a signaling group that contains the registrar table. Deny. Call routes to a specific Q.850 cause code are rejected. Through the Deny Q.850 Cause Code field, select the specific Q.850 Cause code. When Deny is selected, the Deny Q.850 Cause Code field is displayed. Trunk Group. Calls are routed to an incoming trunk group destination using the associated signaling group. This routing entry should be selected in order to route calls to a trunk group. When Trunk Group is selected, Fork Call and Destination Signaling Groups options are not available. |
| Deny Q.850 Cause Code | Specifies the Q.850 Cause Code for which the call is rejected. Select from the drop-down list. This field is displayed only when Deny is selected as Destination Type. |
| Message Translation Table | Specifies which Message Translation Tableto use. |
| Cause Code Reroutes | Specifies which Cause Code Reroute tableto use. |
| Cancel Others upon Forwarding | Specifies whether or not forked calls should clear when one of the forked calls is forwarded. |
| Fork Call | Specifies whether or not to fork a call if this route is selected |
| Destination Signaling Groups | Specifies the Signaling Groups used as the destination of calls. Helpful Hint
This field presents a multi-select widget when the Add/Edit button is clicked. |
| Enable Maximum Call Duration | Enables a maximum time (set in the Maximum Call Duration field) in which a call can stay in the connected state. This field provides another safeguard against stuck calls (i.e., if a call is up for two days and most likely not meant to stay connected). Valid entry: Enabled (any call using that particular call route will disconnect after the configured duration in the connected state) or Disabled (default). |
| Maximum Call Duration | The maximum time (in minutes) in which a call can stay in the connected state. This field is available only when the Enable Maximum Call Duration is set to Enabled. Valid entry: 1 - 10080 minutes. |
| Audio Stream Mode | Media Mode enables you to choose how the SBC handles call signaling and Audio/Fax media stream during a call. Five options are available: DSP (default entry). The SBC uses DSP resources for media handling (transcoding) but it does not facilitate the capabilities/features between endpoints that are not supported within the SBC (codec/capability mismatch). When DSP is configured, the Signaling Groups enabled to support DSP are attempted in order. To enable a Signaling Group for DSP mode, see Creating and Modifying SIP Signaling Groups. Proxy. The SBC will proxy or switch the media stream between endpoints and let the endpoints negotiate common media capabilities and handle unsupported/unknown audio codecs. The media flows through the SBC without using DSP resources. For example, this mode is used when both inbound and outbound calls are SIP and both Signaling Groups are enabled for Proxy mode. If one Signaling Group has Proxy mode disabled or one of the inbound Signaling Groups is TDM, the call is rejected. Route configuration does not allow Destination Signaling Groups of TDM type if Stream Mode is set to Proxy. The failover mechanism can then be used to re-route to another Signaling Group. We strongly recommend using Proxy preferred over DSP mode since it allows the same routing entry with different Signaling Groups. DSP preferred over Proxy. The SBC prefers to use DSP resources, rather than proxy or switch the media stream between endpoints. This mode is chosen during the call setup based on the call routing, and used for call routes where the inbound call route is both TDM and/or SIP (this avoids duplicate call routes). If the inbound Signaling Group is a SIP Signaling Group not enabled for DSP, then Proxy is used. NOTE: If RTP encryption/decryption is expected, it is necessary to set the Proxy SRTP Handling field to 'Local', so that encryption/decryption is supported when switching to proxy. Proxy preferred over DSP. The SBC prefers to proxy the media stream between endpoints rather than choosing DSP mode (which uses the DSP resource). This mode is chosen during the call setup based on the call routing result, and used for call routes where the inbound call route is both TDM and/or SIP (this avoids duplicate call routes). If the inbound or outbound Signaling Group is NOT in Proxy mode (either the SIP signaling group is not enabled for Proxy mode or the Signaling Group is TDM), then DSP is used. Direct. The SBC media flows directly between compatible end-points without using either the SBC processor or a DSP resource. LMO. Use Local Media Optimization (LMO) to route media directly between the Teams client and the downstream SBC. This option attempts to keep media on the LAN side of the network when the Teams client is on the LAN side. When the Teams client is on the WAN side, the media passes through the Proxy SBC and uses the DSP resources. Using this option also reduces the Proxy SBC load, as the proxy SBC is not in the media path when the Teams clients are on the LAN side. Select the LMO option for call routes between the proxy and downstream in both directions. Dynamic DSP. Use the Dynamic DSP as an alternative to the standard DSP. When this option is selected, the SBC SWe Edge attempts to negotiate a common codec between the two endpoints involved in the call to avoid using transcoding, which is CPU-intensive. This results in a higher call capacity than the DSP option, which always uses DSP resources. Note
The DSP resource and transcoding licenses are required when configuring the FAX/MODEM tone detection or DTMF interoperability. LMO with Dynamic DSP: When using this option, the Proxy negotiates a common codec between Teams and Downstream to avoid using DSP resources when the Teams client is on the WAN side of the network. This option works like Dynamic DSP when the Teams client is on the WAN side and LMO when the Teams client is on the LAN side. Note
This option only applies to the SBC SWe Edge. Disabled. The Audio/Fax media selection process is disabled. When Disabled is selected, the SBC will reject incoming Audio/Fax streams. |
| Video/Application Stream Mode | Media Mode enables you to choose how the SBC handles call signaling and Video/Application media stream during a call. Proxy. The SBC will proxy the media stream between endpoints. The media flows through the SBC without requiring DSP resources. For example, this mode is used when an inbound SIP call includes Video/Application streams and both inbound and outbound SIP Signaling Groups are enabled for Proxy mode. If one Signaling Group has Proxy mode disabled, the stream is disabled. If the outbound Signaling Group is TDM, the call will be connected using Audio/Fax media only and Video/Application media stream will be rejected. Direct: The SBC media flows directly between compatible end-points without using either the SBC processor or a DSP resource. Disabled. The Video/Application Stream Mode is disabled. The SBC will not proxy or switch media streams between endpoints. |
| Proxy SRTP Handling | Note
Available for SBC SWe only. Note
SIP Signaling Sessions and Enhanced Media Sessions without Transcoding licenses are required for this functionality. The Proxy SRTP Handling option applies SRTP media encryption to a call; this enables communication between end points that are SRTP incompatible. For example, this enables communication if both Calling and Called user agents both support G.711u codec, but they differ in SRTP capability, or one party in the call does not support SRTP at all. Valid entries: Relay or Local. Default: Relay. This field is displayed only when Proxy, DSP preferred over Proxy, or Proxy preferred over DSP is selected from the Audio Stream Mode drop-down list. Relay: When Relay is selected, the following options are supported:
Local: When Local is selected, SRTP is negotiated automatically for the following:
|
| Media Transcoding | Specifies whether or not to use media transcoding. Media Transcoding requires a specific Transcoding License (Enhanced Media Sessions with Transcoding). Do not enable Media Transcoding unless your calling configuration requires it and SBC Edge Portfolio is licensed for the Transcoding feature. |
Media List | Specifies the Media List to use for this call route. This drop down list is populated with the Media List entries created through the Create Media list option. See Creating and Modifying Media Lists. If the Media List configuration selected, then the Destination Signaling Group would be selected that has the common media set available. The media order from the call route's media list takes precedence over the Signaling Group's media list. Generally this field should be kept the default value "None" unless the media codec selection has to be controlled and manipulated for this route. |
| Quality Metrics Number of Calls | Specifies the number of calls over which the quality metrics are calculated. |
| Quality Metrics Time Before Retry | Specifies the period of time in minutes after which a route is tried again after failing quality metrics. |
| Min. ASR Threshold | Specifies the minimum answer/seizure ratio for this rule to be considered for use. |
| Enable Min MOS Threshold | This field enables the Min. MOS score feature (call routes checked for voice quality) to be used. Valid entries: Enabled (enables the Min. MOS Score feature) or Disabled (disables Min. MOS Score feature) |
| Min. MOS Score | The Min. MOS score provides an option to allow the SBC to temporarily enable/disable a route that is having bad voice quality. Two options are available: Enabled. A MOS score for quality purposes is added to each call using that specific route. The Quality Metrics Number of Calls field is used to determine the average acceptable value; if the average of acceptable calls falls below, it is considered bad voice quality and the route is then temporarily disabled for calls. The Quality Metrics Time Before Retry field is used when the route is disabled for failing metrics to determine how long before the route is retried. Disabled. The Min. MOS Score feature is disabled and all call routes are enabled. |
| Enable Max. R/T Delay | Specifies whether or not to use Round Trip Delay. |
| Max. R/T Delay | Specifies the maximum average round trip (R/T) delay for this rule to be considered for use. |
| Enable Max. Jitter | Specifies whether or not Jitter will be considered as a quality metric for this Call Route. |
| Max. Jitter | Specifies the maximum average jitter for this rule to be considered for use. |
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