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This document outlines the configuration best practices for Ribbon Edge 8K when deployed with Cisco Unified CM and Avaya IPO.
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,
The 8100/8300 platform is based on Intel Atom 8 core processor with multiple interfaces. This platform shall meet following high level requirements :-
Functionality
For additional information on the Ribbon SBC , refer to https://ribboncommunications.com/
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.
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.
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.
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:
This configuration guide is offered as a convenience to Ribbon customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS.” Users must take full responsibility for applying the specifications and information in this guide.
The following aspects are required before proceeding with the interop:
The sample configuration in this document uses the following equipment and software:
Product | Appliance/ Application/ Tool | Software Version |
---|---|---|
Ribbon Communications | Ribbon Edge 8K | V23.06.0 |
Third-Party Products | Cisco Unified Communications Manager | 12.5.1.11900-146 |
Avaya IP Office | V10.1.0.2.0 Build2 | |
Third-party Phones | Cisco CP-8865 VOIP Phones | sip8845_65.12-5-1SR3-74 |
Beetel Analog Phone | - |
Open a browser and enter the SBC Edge IP address.
Click on Enter and then log in using admin credentials.
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
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.
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.
When the Verify Status field in the Certificate panel indicates Expired or Expiring Soon, replace the Trusted CA Certificate. You must delete the old certificate before importing a new certificate successfully.
Most Certificate Vendors sign the SBC Edge certificate with an intermediate certificate authority. There is at least one, but there could be several intermediate CAs in the certificate chain. When importing the Trusted Root CA Certificates, import the root CA certificate and all Intermediate CA certificates. Failure to import all certificates in the chain causes the import of the SBC Edge certificate to fail. Refer to Unable To Get Local Issuer Certificate for more information.
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
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).
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
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
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
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
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 to create a new SRTP profile.
Use the following steps to complete the configuration:
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.
PRI_SG
FXS_SG
CUCM_SG
You can configure SIP Trunk between the service provider and IP-PBX over UDP, TCP, or TLS. Ribbon recommends the use of TLS protocol to ensure security. Customers who do not wish to use TLS as a preferred protocol can skip this section.
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
For details on Transformation Table Entry configuration, refer to Creating and Modifying Entries to Transformation Tables. For call digit matching and manipulation through regular expressions, refer to Creating Call Routing Logic with Regular Expressions.
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
In the call routing table, Audio Stream Mode is, by default, in DSP mode. It is recommended to use the default DSP mode configuration.
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.
Customers are free to choose any transport medium depending on their requirements. Ribbon strongly recommends using secure TLS protocol.
For more information regarding CSR and Certificate generation for CUCM, refer to https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/215412-configure-sip-tls-between-cucm-cube-cube.html
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.
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.
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.
Use a trunk device to configure a logical route to a SIP network.
Resetting/restarting a SIP device does not physically reset/restart the hardware. It only reinitializes the configuration loaded by Cisco Unified Communications Manager.
For SIP trunks, Restart and Reset behave the same way, so all active calls disconnect when either choice is pressed.
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.
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.
CUCM supports auto registration of Cisco endpoints. Refer to the following link for more details:
In Cisco Unified Communications Manager Administration, use the
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.
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".
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.
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.
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.
The following checklist depicts the set of services/features covered through the configurations defined in this Interop Guide.
01. | OPTIONS validation | |
02. | Call Setup and Termination over TLS | |
03. | Ringing and Local Ringback Tone | |
04. | Remote Ringback Tone Handling | |
05. | Cancel Call, No Answer, Busy and Call Rejection | |
06. | Basic Call with different codecs | |
07. | DTMF | |
08. | Anonymous Calls | |
09. | Call Hold and Resume | |
10. | Call Forward - Unconditional, Busy and No Answer | |
11. | Call Transfer (Blind/Unattended) | |
12. | Call Transfer (Attended) | |
13. | Call Conference | |
14. | Meet Me Conference | |
15. | 4xx/5xx Response Handling | |
16. | Long Duration Calls | |
17. | Early and Late Media | |
18. | Simultaneous Ringing | |
19. | Transcode Calls |
Legend
Supported | |
Not supported |
Note the following items concerning this Interoperability. These are limitations or test observations on this Interoperability.
For any support-related queries about this guide, please contact your local Ribbon representative or use the details below:
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
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.