Table of Contents
This document provides a configuration guide for Ribbon Session Border Controller Edge Series (SBC) when connecting to Cisco Unified Communication Manager 11.0 (CUCM 11).
- For additional information on the Cisco Platform, visit http://www.cisco.com.
- For additional information on the Ribbon SBC, visit http://ribboncommunications.com.
The interoperability compliance testing focuses on verifying inbound and outbound call flows between the Ribbon SBC Edge and Cisco Unified Communication Manager 11.0 (CUCM 11).
This technical document is intended for telecommunication engineers with the purpose of configuring the Ribbon SBC Edge series aspects of the AT&T Flex Reach SIP trunk group with the Cisco Unified Communication Manager 11. This configuration requires navigating a third-party server and the Ribbon SBC Web browser user interface, Embedded Management Application (EMA). Understanding the basic concepts for IP/Routing, SIP, RTP, and TLS are also required for completing the configuration and any necessary troubleshooting.
The following equipment and software were used for the sample (see Topology):
Ribbon SBC Edge (2000)
|Cisco UCM 11.0||126.96.36.19900-11|
|Cisco IP Phone 7942|
The following reference configuration illustrates the connectivity between a third-party and the Ribbon SBC Edge.
For any questions regarding this document or the content herein, please contact your maintenance and support provider.
Third-Party Product Features
The testing was executed with the AT&T test plan, and the following features were tested:
- Basic originated and terminated calls
- Calling Number presentation
- Hold and Resume
- Voice Mail
- Conference Call
- Call Transfer
- Call Forwarding
- Auto Attendant
- Meet-Meet Conference
- AT&T IP Teleconferencing
- N11 Calls
- Network Based Enhanced Features
Not Supported Features
- SBC does not send SIP with SDP without p-time
- SBC does not support network based transfer with SIP Refer method
- CUCM does not support SIP REFER method for network transfer
- Voice mail is not supported on the single server deployment.
- PBX-Based Auto Attendant is not supported on the single server deployment.
No special licensing required.
Cisco UCM 11 Configuration
The following new configurations are included in this section:
1. SIP Profile
Select Device > Device Settings > SIP Profile
2. SIP Trunk Security Profile
Select System> Security > SIP Trunk Security Profile
Select Device > Trunk
4. Route Group
Select Call Routing > Route/Hunt > Route Group
5. Route List
Select Call Routing > Route/Hunt > Route List
6. Route Pattern
Select Call Routing > Route/Hunt > Route Patterns
Use this procedure to create any Route Pattern configuration.
6. Meet-Me Number
Select Call Routing > Meet-Me Number
Ribbon SBC Edge Series Configuration
The following steps provide an example of how to configure Ribbon SBC Edge.
- Easy Config Wizard
- SIP Profile
- Q.850 Cause Code to SIP Override Tables
- Tone Table
- Media Profile
- Media System Configuration
- Media List
- Message Manipulation
- SIP Server
- Signaling Group
1. Easy Config Wizard
The SBC interface includes an Easy Configuration Wizard, which enables end users to quickly deploy SBC. Based on a template, you can configure items such as endpoint (define user and provider), routing (routing configuration applied to scenario), and country (tone table parameters and emergency numbers for a particular country).
2. SIP Profile
SIP Profiles control how the Ribbon SBC Edge communicates with SIP devices. The SIP Profiles control characteristics such as
- Session timers
- SIP Header customization
- SIP timers
- MIME payloads
- Option tags
To configure the SIP Profiles, select Settings > SIP > SIP Profiles.
3. Q.850 Cause Code to SIP Override Table
By default, the SBC Edge uses RFC 3398 cause code mappings. Q.850 Cause Code to SIP Override Table allows you to define other mappings for cause codes.
To configure the Q.850 Cause Code to SIP Override Table, select Q.850 Cause Code to SIP Override Tables.
4. Tone Tables
Tone tables allow the SBC Edge administrator to customize the tones a user hears when placing a call. You can modify the tone to match your local PSTN or PBX. The default tone table configures the following categories with the United States' values:
- Call Waiting
To configure the Tone Tables, select Settings > Tone Tables.
5. Media Profile
Media profiles specify the individual voice and fax compression codecs, and their associated settings for inclusion into a Media list. Different codecs provide varying levels of compression, which enables the reduction of bandwidth requirements at the expense of voice quality.
To access the Media Profile, select Settings > Media > Media Profiles.
6. Media System Configuration
The Media System Configuration contains system wide settings for the Media System. To configure the Media System, set the number of RTP/RTCP port pairs and the starting port.
To access the Media Profile, select Settings > Media > Media System Configuration.
7. Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
To access Media lists, select Settings > Media > Media List.
8. Message Manipulation
Condition rules are rules that apply to a specific component of a message (for example, diversion.uri.host, from.uri.host, and such) with the value in the Match Type list box. The value is matched against a literal value, token, or REGEX.
To configure Message Manipulation, select Settings > SIP > Message Manipulation > Condition Rule Table.
The rule on the next figure changes a host part for the PAID (P-Asserted-Identity) header for all outbound calls to ATT SIP Trunk with an IP address of public interface.
9. SIP Server
The SIP Server tables contain information about the SIP devices connected to the Ribbon SBC Edge. The table entries provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting.
To configure the SIP Server, select Settings > SIP > SIP Server Tables.
10. Signaling Group
Signaling Groups allow telephony channels to be grouped together for routing and shared configuration. The Signaling Groups are the entity to which calls are routed and where the Call Routes are selected. In the case of SIP, Signaling Groups will specify protocol settings and link to server, media, and mapping tables.
To configure Signaling Groups, select Settings > Signaling Groups.
Transformation tables facilitate the conversion of names, numbers, and other fields when routing a call. For example, transformation table converts a public PSTN number into a private extension number or a SIP address (URI). Every entry in a Call Routing table requires Transformation tables, which are sequentially selected. In addition, Transformation tables are configurable as a reusable pool that action sets can reference.
To configure the Transformation table, select Settings > Transformation.
12. Call Routing Table
Call Routing allows calls to be carried between Signaling Groups, which allows calls to be carried between ports and between protocols (for example, ISDN to SIP). Routes are defined by Call Routing tables, which allows for flexible configuration of calls that are carried, as well as how the calls are translated. These tables are one of the central connection points of the system linking Transformation tables, Message translations, Cause Code Reroutes, Media lists, and the three types of Signaling Groups: ISDN, SIP, and CAS.
To configure the Call Routing Table, select Settings > Call Routing Table.
These Application Notes describe the configuration steps required for Ribbon SBC Edge Series to successfully interoperate with AT&T IP Flex Reach SIP Trunk. All feature and serviceability test cases were completed.