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Interoperable Vendors

Copyright

© 2023 Ribbon Communications Operating Company, Inc. © 2023 ECI Telecom Ltd. All rights reserved. The compilation (meaning the collection, arrangement and assembly) of all content on this site is protected by U.S. and international copyright laws and treaty provisions and may not be used, copied, reproduced, modified, published, uploaded, posted, transmitted or distributed in any way, without prior written consent of Ribbon Communications Inc.

The trademarks, logos, service marks, trade names, and trade dress (“look and feel”) on this website, including without limitation the RIBBON and RIBBON logo marks, are protected by applicable US and foreign trademark rights and other proprietary rights and are the property of Ribbon Communications Operating Company, Inc. or its affiliates. Any third-party trademarks, logos, service marks, trade names and trade dress may be the property of their respective owners.  Any uses of the trademarks, logos, service marks, trade names, and trade dress without the prior written consent of Ribbon Communications Operating Company, Inc., its affiliates, or the third parties that own the proprietary rights, are expressly prohibited.

Document Overview

This document outlines the configuration best practices for Ribbon Edge 8K when deployed with Cisco Unified CM and Avaya IPO.

About Ribbon Edge 8000

Ribbon’s Edge 8000 is the newest, high-performance member of our line of services gateway routers that combines security, routing, switching, and 10 Gbps WAN interfaces with next-generation voice and data services where the combination of broadband connectivity and advanced threat mitigation capabilities are required.  By consolidating fast, highly available routing, security, and next-generation SBC capabilities in a single device, enterprises can remove network complexity, protect and prioritize resources, and improve user and application experience while lowering total cost of ownership (TCO).

The Edge 8000 series is comprised of two models,

• Edge 8100, a highly scalable Ethernet SBC/data router.
• Edge 8300, a high-capacity mixed SBC/analog/data router.
  
  

The 8100/8300 platform is based on Intel Atom 8 core processor with multiple interfaces. This platform shall meet following high level requirements :-

Functionality

• SBC with or without transcoding
• Support for legacy Analog interfaces (Edge8300 only) - Enables the cost, performance, and availability benefits by collapsing edge routing functionality with legacy termination services in one multiservice box. 
• Routing and Security

About Cisco Unified CM

Cisco Unified Communications Manager (CUCM) is the core call control application of Cisco's collaboration portfolio. It provides reliable, highly secure, scalable, and efficient enterprise call and session management.

About Avaya IP Office

Avaya IP Office (IPO) is a single, stackable, scalable small business communications system that offers technical flexibility using digital (ISDN), analog (FXS), IP (SIP) or any combination of these - and resiliency. The Avaya IP Office Platform is a cost-effective telephony system that supports a mobile, distributed workforce with voice and video on virtually any device.

Scope/Non-Goals

It is not the goal of this guide to provide detailed configurations that will meet the requirements of every customer. Use this guide as a starting point and build the SBC configurations in consultation with network design and deployment engineers. 

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Ribbon SBCs and the third-party product.

To perform this interop, you need to:

• use the graphical user interface (GUI) or command line interface (CLI) of the Ribbon product
• have an understanding of the basic concepts of TCP/UDP/TLS, IP Routing, TDM (FXS/T1-E1/PRI)
• have an understanding of SIP/RTP, and SIP/SRTP to complete the configuration and troubleshoot.

Prerequisites

The following aspects are required before proceeding with the interop:

• Ribbon Edge 8K
• Ribbon Edge 8K license
  • A valid license from Ribbon is required to enable functionality on Ribbon SBCs. Each SBC license provides a base set of capabilities to which users can add and enable additional features as required.
• Cisco Unified Communication server 
  • Cisco Unified Communication server needs to be configured with appropriate licenses. Refer to https://www.cisco.com for more information about licenses.
• TLS Certificates for SBC SWe Edge
  • For more information on TLS certificates, refer to Working with Certificates.

Product and Device Details

The sample configuration in this document uses the following equipment and software:

Network Topology Diagram

Deployment Topology

Interoperability Test Lab Topology

Signaling and Media Flow

Anchor
Section B Section B

Edge 8K Configuration

SBC SWe Edge Configuration

Installation Procedure

• Installation of Edge 8K
• To deploy an Edge 8K instance, refer to https://wiki.rbbn.com/display/EEMPROJ/ISO+installation

Accessing the SBC Edge

Open a browser and enter the SBC Edge IP address.

Click on Enter and then log in using admin credentials.

View License

This page describes how you can view the status of each license along with a copy of the license keys installed on your SBC.

Navigate to System > Licensing > Current Licenses

Import Trusted Root CA Certificates

A Trusted CA Certificate is a certificate issued by a trusted certificate authority. Trusted CA Certificates are imported to the SBC SWe Edge to establish its authenticity on the network.

From the Settings tab, navigate to Security > SBC Certificates > Trusted CA Certificates.

This section describes the process of importing Trusted Root CA Certificates using either the File Upload or Copy and Paste methods.

1. To import a Trusted CA Certificate, click the Import Trusted CA Certificate () Icon.
2. Select either Copy and Paste or File Upload from the Mode menu.
3. If you choose File Upload, use the Select File button to find the file.
4. Click OK.

Use the steps above to import the Service Provider's Root and Intermediate certificates of their Public CA.

For more details on Certificates, refer to Working with Certificates.

Configure Networking Interfaces

This section contains information about how to manage the way the Ribbon SBC SWe Lite interfaces with the network. The SBC SWe Lite supports system-created logical interfaces (Administrative IP, Ethernet 1 IP, Ethernet 2 IP, Ethernet 3 IP). In addition to the system-created logical interfaces, the SBC SWe Lite supports user-created VLAN logical sub-interfaces.

Configure the interface IPs for SBC SWe Lite as follows:

Navigate to Networking Interfaces > Logical Interfaces

Configure Static Routes

Static routes are used to create communication to remote networks. In a production environment, static routes are mainly configured for routing from a specific network to another network that can only be accessed through one point or one interface (single path access or default route).

• For smaller networks with just one or two routes, configuring static routing is preferable. This is often more efficient since a link is not being wasted by exchanging dynamic routing information.
• For networks that have a LAN-side gateway on Voice VLAN or Multi-Switch Edge Devices (MSEs) with voice VLAN towards the SBC Edge, static routing configurations are not required.

Destination IP

Specifies the destination IP address

Mask

Specifies the network mask of the destination host or subnet. If the 'Destination IP Address' field and the 'Mask' field are both 0.0.0.0, the static route is called the 'default static route'.

Gateway

Specifies the IP address of the next-hop router to use for this static route.

Navigate to Protocols > IP > Static Routes

Configure a Local Registrar

SIP provides a registration function that allows users to upload their current locations for use by proxy servers. Registration creates bindings in a location service for a particular domain that associates an address-of-record URI with one or more contact addresses.

Registration entails sending a REGISTER request to a special type of User-Agent Server (UAS) known as a registrar. A registrar acts as the front-end to the location service for a domain, reading and writing mappings based on the contents of REGISTER requests.

In this interop, the FXS endpoints are registered.

Navigate to SIP > Local Registrars

Configure a SIP Profile

SIP Profiles control how the SBC Edge communicates with SIP devices. They control important characteristics such as session timers, SIP header customization, SIP timers, MIME payloads, and option tags.

Navigate to SIP > SIP Profiles

Configure TLS profile

1. Navigate to Security > TLS Profiles
2. Configure TLS profile.

Configure SIP Server Tables

SIP Server Tables contain information about the SIP devices connected to the SBC Edge. The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each server.

Three SIP Server tables are used in this interop.

Navigate to SIP > SIP Server Tables 

Create SRTP Profile

SDES-SRTP Profiles define a cryptographic context that is used in SRTP negotiation. SDES-SRTP Profiles are required to enable encryption, and SRTP is applied to Media Lists. SDES-SRTP Profiles was previously named Media Crypto Profiles.

From the Settings tab, navigate to Media > SDES-SRTP Profiles. Click the  icon to create a new SRTP profile.

Use the following steps to complete the configuration:

1. Provide the desired description for the profile.
2. Set the Operation Option as Required. This setting permits call connections only if you can use encryption for the call. If the peer device does not support SRTP (Secure Real Time Protocol) for voice encryption over the IP network, the call setup will fail.
3. Attach the Crypto suite "AES_CM_128_HMAC_SHAI_80": A crypto suite algorithm that uses the 128-bit AES-CM encryption key and a 80-bit HMAC_SHA1 message authentication tag length.
4. Key Identifier Length is set to "0." Set this value to 0 to disable the MKI in SDP.
5. Click OK.

Media List

Navigate to Media > Media List

Dead Call Detection is accomplished by monitoring incoming RTCP packets. If this feature is enabled and no RTCP packets are received from the peer for 30 seconds, the call is considered "dead" and is disconnected.

• Enable DCD from the options provided in the drop-down

Configure Signaling Groups

1. Signaling groups allow telephony channels to be grouped together for routing and shared configuration. They are the entity to which calls are routed, as well as the location from which Call Routes are selected. 
2. Navigate to Signaling Groups (Add SIP SG)

PRI_SG

• In the SIP Profile, choose the Default SIP Profile.

FXS_SG

1. In the SIP Profile, choose the Default SIP Profile.
2. In Registrar, choose the "Local_register_table" created earlier.
3. In Media List ID, choose the "RTP_LIST" created earlier.
4. Configure required fields as shown.

CUCM_SG

1. In the SIP Profile, choose the Cisco Profile.
2. In Media List ID, choose the "SRTP_LIST" created earlier.
3. Configure required fields as shown.

Configure Call Routing Transformation Tables

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. They can, for example, convert a public PSTN number into a private extension number or into a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table, and they are selected from there.

Navigate to Call Routing > Transformation 

• Configure the required transformation table as shown.

Configure Call Routing Tables

Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried and how they are translated.

Navigate to Call Routing > Call Routing Table 

• Attach the Transformation Table entry created in the previous step.
• Select the Destination Signaling Group as required.

Edge8K Platform Configuration

Accessing the Yang Portal

• Open a browser and enter the Yang GUI IP address.
• Provide the required username and password to log in.

Analog SIP-User-Agent configuration

• Configure and enable the required fields for SIP-User-Agent.

• Configure the required Port-Level-Configuration fields for the phones.

PRI-Setting Configuration

• Configure and enable the required fields for PRI-Setting.

Anchor
Section C Section C

CUCM Configuration

Accessing CUCM (Cisco Unified CM Administration)

1. Open Browse and enter the CUCM IP Address.
2. Select Cisco Unified CM Administration from the Navigation drop-down.
3. Provide the credentials and click Login.

Configure SIP Trunk Security Profile

Unified Communications Manager Administration groups security-related settings for the SIP trunk to allow you to assign a single security profile to multiple SIP trunks. Security-related settings include device security mode, digest authentication, and incoming/outgoing transport type settings.

1. From Cisco Unified CM Administration, navigate to System > Security > SIP Trunk Security Profile.
2. Click Add New.
   • Provide the desired Name and Description.
   • Choose Secure from Device Security Mode.
   • From Incoming Transport Type, select TLS.
     • When Device Security Mode is Non Secure, TCP+UDP specifies the transport type.
   • Select Outgoing Transport Type as TLS.
   • Click Save.

Configure SIP Profiles

A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include name, description, timing, retry, call pickup URI, etc. The profiles contain some standard entries that you cannot delete or change.

1. From Cisco Unified CM Administration, navigate to Device > Device Settings > SIP Profile.
2. Click Add New.
3. Enter a name to identify the SIP profile.
4. Provide a description to identify the purpose of the SIP profile.
5. From the SIP Rel1XX Options drop-down, choose Send PRACK for all 1xx Messages.
6. From the Early Offer support for voice and video calls drop-down, choose Best Effort (no MTP inserted).
       - Provide an Early Offer for the outbound call only when the caller's media port, IP, and codec information is available.
       - Provide a Delayed Offer for the outbound call when the caller's media port, IP, and codec information are unavailable. No MTP is inserted to provide an Early Offer in this case.
7. Enable SIP OPTIONS Ping.
       - SIP OPTIONS are requests to the configured destination address on the SIP trunk.
8. Click Save.

Configure Media Resource Group 

Media resource management comprises working with media resource groups and media resource group lists. Media resource management provides a mechanism for managing media resources so all Cisco Unified Communications Managers within a cluster can share them. Media resources provide conferencing, transcoding, media termination, annunciator, and music on hold services.

1. From Cisco Unified CM Administration, navigate to Media Resources > Media Resource Group.
2. Click Add New.
3. Enter a unique name in this required field to identify the media resource group.
4. Enter a description for the media resource group. 
5. To add a media resource for this media resource group, choose one (MoH_2 in this case) from the available Media Resources list and click the down arrow. After a media resource is added, its name moves to the Selected Media Resources pane.
6. Click Save.

Configure Media Resource Group  List

A Media Resource Group List provides a prioritized grouping of media resource groups. An application selects the required media resource, such as music on hold server, from among the available media resources according to the priority order defined in a Media Resource Group List.

1. From Cisco Unified CM Administration, navigate to Media Resources > Media Resource Group List menu path to configure media resource group lists.
2. Click Add New.
3. Enter a unique name in this required field to identify the Media Resource Group List.
4. Choose the Media Resource Group created in the previous step from the Available Media Resource Groups list and click the down arrow between the two panes. After a media resource group is added, its name moves to the Selected Media Resource Groups pane.
5. Click Save.

Trunk Configuration

Use a trunk device to configure a logical route to a SIP network.

1. From Cisco Unified CM Administration, choose Device > Trunk.
2. Click Add New.
3. From the Trunk Type drop-down list, choose SIP Trunk.
4. Choose SIP from Device Protocol drop-down.
5. From the Trunk Service Type, select the default value (None).
6. Click Next.
7. Enter a unique identifier for the trunk.
8. Enter a descriptive name for the trunk.
9. Choose the Default Device Pool.
10. Choose the Media Resource Group List created in the previous step.
11. Provide the destination address.
        - The Destination Address represents the remote SIP peer with that this trunk will communicate.
        - SIP trunks only accept incoming requests from the configured Destination Address and the incoming port that is specified in the SIP Trunk Security Profile that is associated with this trunk.
12. Choose the SRTP Allowed (only when the SIP Trunk profile is created as TLS)
13. Choose the SIP Trunk Security Profile created to apply to the SIP trunk.
14. Select the SIP Profile created from the list.
15. Choose RFC 2833 as the DTMF Signaling Method.
16. Click Save.
17. Reset, Restart and Close the window. Refresh the SIP trunk page and wait until the Server status changes from Unknown to Full Service.

Configure Call Routing

A route pattern comprises a string of digits (an address) and a set of associated digit manipulations that route calls to a route list or a gateway. Route patterns provide flexibility in network design. They work in conjunction with route filters and route lists to direct calls to specific devices and to include, exclude, or modify specific digit patterns.

1. In Cisco Unified Communications Manager Administration, use the Call Routing > Route/Hunt > Route Pattern menu path to configure route patterns.
2. Click Add New.
3. Enter the route pattern, including numbers and wildcards (do not use spaces); for example, for NANP, enter 9.@ for typical local access or 8XXX for a typical private network numbering plan. Valid characters include the uppercase characters A, B, C, and D and \+, representing the international escape character +.
4. Configure the Route Pattern as below.
5. Choose SIP Trunk created from the gateway or route list drop-down to add the route pattern. 
6. Click Save.

Configure End Users

The End User Configuration window allows you to add, search, display, and maintain information about Unified Communications Manager end users. End users can control phones after associating a phone in the End User Configuration window. 

1. In Cisco Unified CM Administration, use the User Management > End User menu path to configure end users.
2. Click Add New.
3. We have two examples taken to configure End Users (Cisco Jabber and Cisco DX650).
4. Enter the unique end-user identification name.
5. Enter alphanumeric or special characters for the end user password and confirm the same.
6. Enter numeric characters for the end-user PIN and confirm.
7. Enter the end-user's last name.
8. For Digest Credentials, enter a string of alphanumeric characters and confirm.
9. For Cisco Jabber, enter the information as shown below.

Phone Setup

1. In Cisco Unified Communications Manager Administration, use the Device > Phone menu path to configure phones.
2. Click Add New
3. From the Phone Type drop-down, choose Cisco 8865.
4. Click Next.
5. Choose Device Trust Mode as Not Trusted if the third-party endpoint is selected for the phone button template.
6. Enter the Media Access Control (MAC) address that identifies Cisco Unified IP Phones. Ensure that the value comprises 12 hexadecimal characters.
7. Choose Default Device pool.
       - A Device pool defines common device characteristics, such as region, date/time group, and soft key template.
8. Choose Cisco Unified Client Services Framework for Jabber clients or Cisco DX650 for DX650 phones from the phone button template drop-down.
       - The phone button template determines the configuration of buttons on the phone and identifies which feature (line, speed dial, and so on) is used for each button.
9. Associate the Media Resource Group List created.
10. Choose the user ID of the assigned phone user.
11. Choose the security profile to apply to the device. Customers can choose to have a Non-Secure SIP Profile if they are using a Non-Secure SIP Trunk.
12. Associate the SIP Profile created before.
        - SIP profiles provide specific SIP information for the phone, such as registration and keep-alive timers, media ports, and do not disturb control.
13. Choose an end user that you want to associate with the phone for this setting that is used with digest authentication (SIP security).
14. Update the CAPF information.
15. Click Save.
16. For the DX650, select the options shown below.
17. Click this link to add a remote destination to associate with this device. The Remote Destination Configuration window displays, which allows you to add a new remote destination to associate with this device.
18. Add the Directory number.
19. Click Save.

1. Click the Associate End User button.
2. Select the end user created from the list and click Add Selected.
3. After completing Step 2, the user association is completed.
4. Save the configuration.
5. Click Apply Config, followed by the Reset button.
6. Reset, Restart and Close the window.

Device Association

1. Navigate back to User Management > End User.
2. In the Device Information field, click Device Association. This will display all the available devices. 
3. Select the device created in the previous step and save.
4. After selecting the appropriate device, it will appear in the Controlled Devices pane.

Enable MoH

In Cisco Unified Communications Manager Administration, use the System > Service Parameters menu path to configure service parameters.

1. In the Server drop-down list box in the Service Parameter Configuration window, choose the CUCM server used. In this case, active means that you provisioned the server in Cisco Unified Communications Manager Administration.
2. From the Service drop-down, select Cisco CallManager. The service displays as active in the Service Parameters Configuration window.
3. Set the Duplex Streaming Enabled flag to True. This parameter determines whether Music On Hold (MOH) and Annunciator use duplex streaming.
4. Click Save.

1. From the Service drop-down, select Cisco IP Voice Streaming App. The service displays as active in the Service Parameters Configuration window.
2. Set the Supported MOH Codecs.
3. Click Save.

Certificate Upload 

1. Navigate back to Security > Upload Certificate/Certificate chain.
2. Choose the file and upload the required certificate.

Avaya IP Office Configuration

We used Avaya IPO for ISDN PRI Trunk termination.

The Avaya IP Office Manager was loaded onto the tester’s PC and allowed the user to log in and access the Avaya IP Office PBX. With Avaya IP Office Manager loaded on your local PC, select Program Files (x86) > Avaya > IP Office > Manager. Select the “Manager” application.

ISDN PRI Trunk

To access the System settings, click the name of the IP Office system. Select Sonus IP Office → Line → .5 (configured as PRI Trunk) → PRI 24 Line.

To configure PRI Trunk, open the Avaya Manager. Go to the "Line" section, create a Line and specify the ISDN Physical Port number (which has T1 connected).

In the following sample config, Port number 9 (though Line number is 05) is configured as PRI as that port number is ISDN in equipment.

Switch Type and Clock Quality are changeable according to customer requirements.

Configure the PRI Channels individually as "In Service" or "Out Of Service." The direction is either incoming, outgoing or bothway.

Configure each Channel with the Line Group ID. In the following sample config, it is configured as "52".

POTS Line

Connect one POTS Phone in one of the FXS Port in Avaya IPO. Go to the "Extension" section, create a new extension ID and extension number, and specify the correct Physical Port.

In the following sample config, the POTS phone is connected to Port 2.

Click "Standard Telephone" for a normal POTS Phone.

Outgoing Call Routing

Go to the "Short Code" section, create a new short code and feature "Dial" and Line Group ID.

Line Group ID is an important configuration. Line Group ID must match with the outgoing Trunk's Line Group ID.

In the following sample config, 992xxxx means after 992, dial four more digits and any four digits after 992.

Incoming Call Routing

Go to the Incoming call Route section. Line Group ID "0" means the call can come from any "Line Group ID." Specify the incoming number.

When the incoming number is matched, route the call to the configured "Destination" on the Destination Tab. In this case, Destination is one of the FXS Port (Port 2).

Go to the Destination Tab and select "User" (example: 210 Extn210) configured under the User section with extension "210" configured under the "Extension" section with Port number 2 in the following example.

The "User" section is shown in the following screen capture.

The "Extension" section is shown in the following screen capture.

Port 2 is linked to Extension 210.

Supplementary Services and Features Coverage

The following checklist depicts the set of services/features covered through the configurations defined in this Interop Guide.

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Legend

Caveats

Note the following items concerning this Interoperability. These are limitations or test observations on this Interoperability.

• Meet Me and Adhoc conference could not be tested due to the unavailability of a hardware transcoder within the lab environment. Lab has CUCM software conference bridge that does not support sRTP. Customers using non-secure trunks and media will not face this issue. For more details, visit https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/11_0_1/secugd/CUCM_BK_C1A78C1D_00_cucm-security-guide-1101/secure_conference_resources_setup.pdf
• A one-way audio issue is observed between the endpoints for basic SRTP call flow scenarios, a known issue to Ribbon. This issue has been addressed and will be fixed in the upcoming SBC release.

Support

For any support-related queries about this guide, please contact your local Ribbon representative or use the details below:

• Sales and Support: 1-833-742-2661
• Other Queries: 1-877-412-8867
• Website: https://ribboncommunications.com/services/ribbon-support-portal

References

For detailed information about Ribbon products and solutions, please visit: https://ribboncommunications.com/products.

For additional information on Cisco Unified Communications Manager, please visit:  https://www.cisco.com/c/en/us/support/unified-communications/unified-communications-manager-callmanager/products-installation-and-configuration-guides-list.html

For additional information on Avaya IP Office, please visit: https://documentation.avaya.com/bundle?labelkey=IP_Office

Conclusion

This Interoperability Guide describes a successful configuration of interop involving Ribbon SBC Edge 8K with CUCM & Avaya IPO.. All the necessary features and serviceability aspects are covered per the details provided in this interoperability document.

Configuration guidance is provided to enable the reader to replicate the same base setup - there may be additional configuration changes required to suit the exact deployment environment.

© 2023 Ribbon Communications Operating Company, Inc. © 2023 ECI Telecom Ltd. All rights reserved.