Table of Contents


 

Interoperable Vendors


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Document Overview

This document outlines the configuration best practices for the Ribbon solution covering the Edge series ATA (Analog Telephone Adaptor)/IAD (Integrated Access Device) when deployed with Zoom.

The Ribbon Edge is a low-density analog gateway or ATA/IAD device that gives small businesses, SOHOs and branch offices with analog voice infrastructures an easy, cost-effective way to capitalize on Voice over Internet Protocol​ (VoIP) services. The Ribbon Edge ATA/IAD offers a survivable branch office solution, providing support for integrating analog endpoints and the Public Switched Telephone Network (PSTN) and support for all Session Initiation Protocol (SIP) calls. The Ribbon Edge supports any-to-any connectivity between analog and SIP devices, enabling branch offices to rapidly migrate analog phones onto SIP-based networks and communicate seamlessly.

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon Edge ATA & Zoom Cloud.

This guide contains the following configuration sections: 

Non-Goals

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

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring the Ribbon SBC.

To perform this interop, you need:

  • to use graphical user interface (GUI) or command line interface (CLI) of the Ribbon product.
  • to understand the basic concepts of TCP/UDP/TLS and IP/Routing.
  • to have SIP/RTP/SRTP to complete the configuration and for troubleshooting.


Note

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 the application of the specifications and information in this guide.

Prerequisites

The following aspects are required before proceeding with the interop:

Product and Device Details

The following equipment and software were used for the sample configuration provided:

Requirements


Equipment

Software Version

Ribbon Communications

Ribbon Edge SBC 1K 

V9.0.4b595

Zoom Zoom Desktop app
5.7.7 (1105)
Zoom Mobile app5.8.1 (2403)

Third-party Phones

Beetel Analog PhoneNA

Note
  • The Ribbon Edge portfolio includes the SBC 1000 and SBC 2000. As a result, this configuration guide is valid for both devices.
  • Zoom Desktop app version is 5.7.7 (1105) or later.
  • Zoom Mobile app version is 5.8.1 (2403) or later.


Network Topology Diagram

This section covers the Ribbon Edge deployment topology and the Interoperability Test Lab Topology.

Ribbon Edge Deployment Topology

Ribbon Edge Deployment Topology

Interoperability Test Lab Topology

The following lab topology diagram shows connectivity between Zoom and Ribbon Edge ATA/IAD.

Interoperability Test Lab Topology

Section A: Ribbon SBC Edge Configuration

The following SBC Edge configurations are included in this section:

  1. Connectivity
  2. Network
  3. Static Routes
  4. TLS Configuration between Ribbon SBC Edge and Zoom

  5. Media Profile

  6. SRTP Profile
  7. SIP Profile

  8. Analog Leg Configuration

  9. Zoom Leg Configuration

1. Connectivity

SBC1K Front Panel


Note

The SBC 1000/2000 is connected to the network as follows:

Ethernet 1: RJ45 "1" is connected towards the PSTN leg.

Ethernet 2: RJ45 "2" is connected towards the Zoom leg.

2. Network

Configure Ethernet 1 and Ethernet 2 of SBC 1000/2000 with the IP as follows:

Navigate to Node Interfaces > Logical Interfaces.

Logical Interfaces

 

Ethernet 1

Ethernet 2

3. 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 a network that can only be accessed through one point or one interface (single path access or default route).

Tip
  • 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.

Add Static routes need to be added towards Eth1 interface 172.16.X.X(PSTN) and Eth2 interface 162.12.X.0(Zoom), as Zoom uses multiple IPs in this subnet.

Default static route is towards the Eth1 that is in a private network.

  • Navigate to Settings > Protocol > IP > Static Routes to configure the routes.

Static Routes

4. TLS Configuration Between Ribbon SBC Edge and Zoom

Prerequisites:

  • For TLS to work on the public side of network, a trusted CA (Certificate Authority) is needed. In this scenario, use GoDaddy as a Trusted CA.
  • Enable Zoom trunk with TLS/SRTP.

Request a certificate for the SBC and configure it based on the example using GoDaddy as follows:

  1. Generate a Certificate Signing Request (CSR) and obtain the certificate from a Certification Authority.
  2. Import the Public CA Root/Intermediate Certificate and SBC certificate on the SBC.

Step 1: Generate a Certificate Signing Request and obtain the certificate from a Certification Authority (CA).

  1. Navigate to Settings > Security > SBC Certificates.
  2. Click Generate SBC Edge CSR.

  3. Enter data in the required fields.

  4. Click OK. After the Certificate Signing request finishes generating, copy the result to the clipboard.

Generate Certificate Signing Request


  5. Use the generated CSR text from the clipboard to obtain the certificate. 

Step 2: Deploy the Root/Intermediate and SBC Certificates on the SBC.

After receiving the certificates from the certification authority, install the SBC Certificate and Root/Intermediate Certificates as follows:

  1. Obtain Trusted Root and Intermediary signing certificates from your certification authority.
  2. To install Trusted Root/Intermediate Certificates, go to Settings > Security > SBC Certificates > Trusted Root Certificates.
  3. Click Import and select the trusted root certificates.
  4. To install the SBC certificate, open Settings > Security > SBC Certificates > SBC Edge Certificate.
  5. Validate the certificate is installed correctly.

Trusted CA certificate table


    6. Click Import and select X.509 Signed Certificate.

    7. Validate the certificate is installed correctly.

Validate certificate

 The following certificates need to be downloaded from Zoom Portal and uploaded into an SBC Edge Under Trusted CA Certificate.

Zoom Certificates


As mentioned in Zoom Portal "Please download DigiCert Global Root CA, DigiCert Global Root G2, DigiCert Global Root G3 and import to your IP phone if they are not in the trust list of the device."


TLS Profile

TLS Profile is required for the TLS handshake between the SBC Edge and Zoom. This profile defines cipher suites supported by the SBC Edge.

Default TLS Profile need to be attached to SIP Server Table on Zoom leg.

Navigate to Security > TLS Profiles. Use the Default TLS Profile with the following modifications:

  • TLS Protocol as "TLS 1.0-1.2 ".
  • Mutual Authentication "Disabled".
  • Certificate as "SBC Edge Certificate".

Default TLS Profile

5. Media Profile

To create a Media Profile:

  • Navigate to Settings > Media > Media Profiles.
  • From the drop-down select Create Media Profile > Voice Codec Profile.

Media Profile

G711-A law

G711 Mu law

6. SRTP Profile

To create a SRTP Profile:

  • Navigate to Settings > Media > SDES-SRTP Profiles.
  • Select the Crypto Suite as "AES_CM_128_HMAC_SHA1_32".
  • Set the LifeTime Value as shown in the diagram.

SDES-SRTP Profile

7. SIP Profile

A SIP profile is used to modify the different SIP parameters like Session timers, SIP Header Customization, and SDP Customization. Create a new SIP Profile "ZOOM_SIP".

  • Navigate to SIP > SIP Profiles > ZOOM_SIP.

SIP profile


8. Analog Leg Configuration

Create profiles with a specific set of characteristics corresponding to Analog (PSTN). This includes configuration of the following entities on the Analog leg:

  1. CAS Profile
  2. Signaling Groups
  3. Transformation
  4. Call Routing Table

1. CAS Profile

  • Navigate to CAS > CAS Signaling Profiles > Create CAS Profile.


CAS Profile


2. Signaling Groups

Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. The signaling groups are the entity where calls are routed, as well as the location where Call Routes are selected. They are also the location where the Tone Tables and Action Sets are selected. 

  • Navigate to Settings > Signaling Groups > Create Signaling Group.
  • From the drop-down select "CAS Signaling Group".
  • Set CAS Signaling Profile as "(FXS)Zoom".
  • Set Tone Table as "Default Tone Table".

  • Assign the Channel Phone Number "809" for Port 1:1.
  • Assign the Channel Phone Number "810" for Port 1:2.
    Tip

    Set Call Routing table as "Analog_Zoom" that is created in the Call Routing Table section.

Signalling Groups

3. Transformation

Transformation table is used to map the 3 digit analog line to 20 digit Zoom Username. In the current setup, "analog_sip" transformation rule applied for outgoing call towards Zoom.

Navigate to Settings > Call Routing > Transformation.

Transformation


4. Call Routing Table

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 where calls are carried and how they are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

  • Navigate to Settings > Call Routing > Call Routing Table.
  • Set Number/Name Transformation Table as "Add_plusOne" as created in an earlier step.
  • Destination Signaling Groups as "(SIP)Zoom_SG" in the Media. Select the following options:
    • Audio/Fax Stream Mode as "DSP".
    • Media Transcoding as "Disabled".
    • Media list as "Default Media List".
Tip

Attach the Media List and Destination Signaling Groups that are created in Zoom Leg Configuration.


Call Routing Table

9. Zoom Leg Configuration

Create profiles with a specific set of characteristics corresponding to Zoom. This includes configuring the following entities on the Zoom leg:

  1. Media List.
  2. Contact Registrant Table.
  3. Remote Authorization Tables.
  4. SIP Server Tables.
  5. Signaling Group.
  6. Transformation.
  7. Call Routing Table.

1. Media List

Media List allows you to specify a set of codecs used for the call. They contain a list of codecs, defined in Media Profile

  • "Add/Edit" to add the different Media profile as created earlier.
  •  As the Zoom leg would be SRTP, attach the SDES-SRTP Profile as "Zoom_SRTP" as created earlier.
  •  Set RTCP mode to "RTCP".
  •  Set Silence Suppression to "enabled".

Media List

2. Contact Registrant Table

The Contact registrant table contains the address of record URI required for REGISTER with Zoom. 

  • Navigate to Settings > SIP > Contact Registrant Table.
  • Type of Address of Record "Remote".
  • Address of Record URI.
  • Add the Contact URI Username.

Contact Registrant Table - Port 1 Config

Port 2 Config


3. Remote Authorization Tables

A Remote authorization table contains a realm, authentication ID and password, used when the REGISTER and INVITE is challenged by Zoom.

  • Navigate to Settings > SIP > Remote Authorization Tables.
  • Realm as "10000587.zoom.us".
  • Authentication ID .
  • From URI User Match "Regex".

Remote Authorization Table - Port 1 Config

Port 2 Config


4. SIP Server Tables

The 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 transport protocols used to communicate with each server. 

  • Navigate to Settings > SIP > SIP Server Tables >Create SIP Server.
  • From the drop-down select "IP/FQDN".
  • Configure the SIP server table with Zoom Fqdn (for example, 10000587.zoom.us in our case).
  • Port as "5091".
  • Configure Transport protocol as "TLS".
  • Set TLS Profile as "ZOOM_TLS" as created in the section TLS Profile.

SIP

 

5. Signaling Groups

Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. The signaling groups are the entity where calls are routed, as well as the location where Call Routes are selected. They are also the location where Tone Tables and Action Sets are selected. In the case of SIP, they specify protocol settings and link to server, media and mapping tables.

  • Navigate to Settings > Signaling Groups > Create Signaling Group.
  • From the drop-down select "SIP Signaling Group".
  • Set SIP Profile as "ZOOM_SIP".
  • Set SIP Server Table as "zoom".
  • Set Media List ID as "Zoom_ML".
  • Set Signaling/Media Source IP as "Ethernet 2 IP(115.110.X.X)".
  • Configure Federated IP as Zoom IP (162.12.X.X).
Tip

Set the Call Routing table as "Zoom_RT" as created in the Call Routing Table section.


Signalling Groups

6. Transformation

The "Remove_plusOne" transformation rule is required for the call towards PSTN.

Navigate to Settings > Call Routing > Transformation.

Transformation


7. Call Routing Table

Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (such as ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration where calls are carried, and how they are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

  • Navigate to Settings >Call Routing > Call Routing Table.
  • Set Number/Name Transformation Table as "Remove_plusOne" as created in an earlier step.
  • Destination Signaling Groups as "(SIP)IP_PBX_SG" In the Media, select the following options:
    • Audio/Fax Stream Mode as "DSP".
    • Media Transcoding as "Enabled".
    • Media list as "Default Media List".
Tip

Attach the Media List and Destination Signaling Groups that are created in the Analog Leg Configuration section.


Call Routing Table


Section B: Zoom Configuration

The SBC 1000 can be provisioned as an ATA/IAD with Zoom, contact Zoom for Provisioning details.

The details provided by Zoom to configure Analog(FXS) Port 1 and Port 2 for the SBC 1000 are as follows:

Ribbon SBC 1K Generic Device Account details
MAC: 00:10:23:e1:04:2f

SIP Account SBC-1K-FXS1
Extension: 809

1. SIP Domain: 10000587.zoom.us
2. Outbound Proxy: gosip0h.sc.zoom.us:5091
3. User Name: 77746483298489165551
4. Authorization ID: 475370019876
5. Password: xxxxx
=============================================

SIP Account 1SBC-1K-FXS2
Extension: 810

1. SIP Domain: 10000587.zoom.us
2. Outbound Proxy: gosip0h.sc.zoom.us:5091
3. User Name: 20407385877802799170
4. Authorization ID: 247646854027
5. Password: xxxxx

Configuring Supplementary Services on Zoom

Zoom supports multiple supplementary services. To configure different supplementary services in Zoom, refer to the following links:

Supplementary Services and Features Coverage

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

Sr. No.Supplementary Features/ServicesCoverage
1Basic Registration over TLS

2Basic Call Setup

3Basic Call Termination

4Auto Receptionist (Auto Attendant)

5Call Hold/Resume

6Call Transfer - Blind (Cold transfer)

7Call Transfer - Consult (Warm transfer)

8Conference

9Call Waiting

10Call Queue

11Shared Line Group (SLG)

12Shared Line Appearance (SLA) or Call Delegation

13Call Recording

14Call Flip

15Call Park

Legend

Supported

Not Supported
N/ANot Applicable

Caveats

The following issues are observed during the testing:

  • Ringback tone not heard on analog phone, when call is dialed from Analog Phone behind SBC 1K/2K to Zoom Client. 
  • The SBC Edge sends 503 Service Unavailable for the Out-of-Dialog NOTIFY received from Zoom Server.

The issues above must fixed in the upcoming SBC 1000 and 2000 software release.

Support

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

References

For detailed information about Ribbon products & solutions, refer to:

https://ribboncommunications.com/products

For information about Zoom products & solutions, refer to:

https://zoom.us/

Conclusion

This Interoperability Guide describes a successful configuration covering Zoom interop with Ribbon Edge ATA/IAD.

All features and capabilities tested are detailed within this document - any limitations, notes or observations are also recorded in order to provide the reader with an accurate understanding of what has been covered, and what has not.

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




© 2021 Ribbon Communications Operating Company, Inc. © 2021 ECI Telecom Ltd. All rights reserved.