Ribbon SBC Edge 1K_2K R11.0 PRI Interop with Google Voice SIP Link : Interoperability Guide

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 SBC Edge 1000 PRI interworking with Google Voice SIP Link. 

About Ribbon SBC Edge 1000

The Ribbon SBC Edge 1000 provides best-in-class communications security. The SBC Edge 1000 dramatically simplifies the deployment of robust communications security services for SIP Trunking, Direct Routing, and Cloud UC services. The SBC 1000 are hardware appliance-based platforms that are part of the Ribbon SBC Edge Portfolio, which addresses the security and interoperability challenges associated with SIP-based communications. The SBC 1000 includes options for Foreign Exchange Office (FXO)/Foreign Exchange Subscriber (FXS) ports and T1/E1 Channel-associated Signaling (CAS)/Primary Rate Interface (PRI) ports. The SBC 1000 is ideally suited for small to medium size organizations and branch offices.

About Google Voice

Google Voice is a telephone service that provides a U.S. phone number to Google Account customers in the U.S., and to Google Works customers in Canada, Denmark, France, the Netherlands, Portugal, Spain, Sweden, Switzerland and the United Kingdom. Calls are forwarded to the phone number that each user must configure in the account web portal. Users can answer and receive calls on any of the phones configured to ring in the web portal. While answering a call, the user can switch between the configured phones. Subscribers in the United States can make outgoing calls to domestic and international destinations. The service is configured and maintained by users in a web-based application, similar in style to Google's email service Gmail, or Android and iOS applications on smartphones or tablets.

Scope/Non-Goals

This document provides configuration best practices for deploying Ribbon's SBC Edge 1000 PRI interop for Google Voice SIP Link. Note that these are configuration best practices and each customer may have unique needs and networks. Ribbon recommends that customers work with network design and deployment engineers to establish the network design which best meets their requirements.  

It is not the goal of this guide to provide detailed configurations that 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 the Ribbon SBC.

To perform this interop, you need to

• use graphical user interface (GUI) or command line interface (CLI) of the Ribbon product.
• understand the basic concepts of TCP/UDP/TLS and IP/Routing.
• understand the basic concepts of T1/E1/ISDN.
• have basic knowledge on SIP/RTP/SRTP to complete the configuration, and for troubleshooting.

Prerequisites

The following aspects are required before proceeding with the interop:

• Ribbon SBC Edge 1000
• Ribbon SBC Edge 1000 license
  
  • This interop requires the acquisition and application of SIP sessions, as documented at Working with Licenses
  • Requires the license for DSI and FXI ports (ISDN ports).
• Public IP addresses
• TLS certificates for SBC Edge 1000
  • For more details, please visit Working with Certificates
• Google Workspace and Domain
  • Google Voice Premier license for the users
  • For more details, contact Google support

Product and Device Details

The configuration uses the following equipment and software:

Network Topology and E2E Flow Diagrams

Deployment Topology

Interoperability Test Lab Topology

Call Flow Diagram

Document Workflow

The sections in this document follow the sequence below. The reader is advised to complete each section for successful configuration.

Installing Ribbon SBC Edge 1000

To deploy the Ribbon SBC Edge 1000 instance, refer to Installing SBC 1000/2000

Ribbon SBC Edge 1000 Configuration

Accessing SBC Edge 1000

Open any browser and enter the SBC Edge 1000 IP address.

Click Enter and log in with a valid User ID and Password.

License and TLS Certificates

View License

This section describes how to view the status of each license along with a copy of the license keys installed on your SBC. The Feature Licenses panel enables you to verify whether a feature is licensed, along with the number of remaining licenses available for a given feature at run-time.

From the Settings tab, navigate to System > Licensing > Current Licenses.

This interop requires license for ISDN ports (DSI/FSX ports).

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

SBC Certificate

From the Settings tab, navigate to Security > SBC Certificates > Generate SBC Edge Certificates.

1. Provide the Common Name of the SBC that includes Host and Domain.
2. Set the Key Length to 2048 bits.
3. Provide the location information.
4. Click OK.
5. The CSR will be generated and displayed in the result text box.

After generating the CSR on Ribbon SBC, provide it to the Certificate Authority. CA would generally provide the following certificates:

• SBC Certificate
• CA's Root Certificate
• Intermediate Certificate

There are two ways to import SBC Primary Certificate as described below:

To import an X.509 signed certificate:

1. Select X.509 Signed Certificate from the Import menu at the top of the page.
2. Chose the import mode (Copy and Paste or File Upload) from the Mode pull-down menu.
3. If you chose File Upload, use the Browse button to find the file and click OK.
4. If you choose Copy and Paste, open the file in a text editor, paste the contents into the Paste Base64 Certificate text field and click OK.

To import a PKCS12 Certificate and Key:

1. Select PKCS12 Certificate and Key from the Import menu at the top of the page.
2. Enter the password used to export the certificate in the Password field.
3. Browse for the PKCS certificate and key file and click OK.

Trusted CA Certificates

A Trusted CA Certificate is a certificate issued by a Trusted Certificate Authority. Trusted CA Certificates are imported to the SBC Edge 1000 to establish its authenticity on the network.

• For TLS to work, a Trusted CA (Certificate Authority) is required. For this interop, GoDaddy is used as the Trusted CA.
• Add an entry in the Public DNS to resolve Ribbon SBC Edge 1000 FQDN to Public IP Address.
• Ensure to have the following certificates as part of the root certificate trust:
  • GTS Root R1
  • GlobalSign Root CA (if required)

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 method.

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.

Follow the steps above to import GTS Root R1 and GlobalSign Root CA certificates from Google Voice.

Networking Interfaces

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

Navigate to Node Interfaces > Logical Interfaces.

Ethernet 1 IP

Ethernet 1 IP is assigned an IP address used for transporting all the VOIP media packets (for example, RTP, SRTP) and all protocol packets (for example, SIP, RTCP, TLS). DNS servers of the customer's network should map the SBC Edge 1000 system hostname to this IP address. In the default software, Ethernet 1 IP is enabled and an IPv4 address is acquired via a connected DHCP server. This IP address is used for performing Initial Setup on the SBC Edge 1000.

Ethernet 2 IP

After initial configuration, you may configure this logical interface using the Settings or Tasks tabs in the WebUI or you can use the IP address configured during Initial Setup.

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 you can only access through one point or one interface (single path access or default route).

Destination IP
Destination IP specifies the destination IP address.

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

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

Metric
Metric specifies the cost of this route and therefore indirectly specifies the preference of the route. Lower values indicate more preferred routes. The typical value is 1 for most static routes, indicating that static routes are preferred to dynamic routes.

From the Settings tab, navigate to Protocols > IP > Static Routes. Click the  icon to add the entries.

Global Configuration

Media Profiles

Media Profiles allow you to specify the individual voice and fax compression codecs and their associated settings, for inclusion in a Media List. Different codecs provide varying levels of compression, allowing one to reduce bandwidth requirements at the expense of voice quality.

From the Settings tab, navigate to Media > Media Profiles. From the Create Media Profile drop-down, select Voice Codec Profile.

The codecs G711A and G711U are configured on the SBC Edge 1000 by default. Configure G722 by following the steps provided below:

For G722:

1. Provide the profile's description.
2. Select G.722 from the Codec drop-down menu.
3. Click OK. 

For G729:

1. Provide the profile's description.
2. Select G.729 from the Codec drop-down menu.
3. Click OK. 

Transformation Table 

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. In addition, Transformation tables are configurable as a reusable pool that Action Sets can reference.

From the Settings tab, navigate to Call Routing > Transformation. Click the  icon to create a Transformation Table.

1. Provide a name for the Transformation Table in the Description field.
2. Click OK.

Transformation Table Entry

1. Click on the Transformation Table created in the previous step.
2. Click the  icon to create an entry.
3. Provide the values in Input and Output fields.
4. Click OK.

Transformation Table Entry for GV_TO_PBX

1. Provide the DID range for PBX as value in the Input Field.
2. Click OK.

Transformation Table Entry for GV_TO_PSTN

1. Provide the DID number range of PSTN as value in the Input Field. Here all 10 digit numbers are allowed.
2. Click OK.

Transformation Table Entry for PBX_TO_GV

1. Provide the DID number range of GV as value in the Input Field. 
2. Click OK.

Transformation Table Entry for PBX_To_PSTN

1. Provide the DID number range of PSTN as value in the Input Field. Here all the Numbers/Address are allowed.
2. Click OK.

Transformation Table Entry for PSTN_TO_GV

1. Provide the DID number range of GV as value in the Input Field.
2. Click OK.

Transformation Table Entry for PSTN_TO_PBX

1. Provide the DID number range of PBX as value in the Input Field.
2. Click OK.

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 flexible configuration of how calls are to be carried and how they are translated. These tables are the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists, and the Signaling Groups.

From the Settings tab, navigate to Call Routing > Call Routing Table. Click the  icon to create a Call Routing Table.

1. Provide a name for the Routing Table in the Description field.
2. Click OK.

Call Routing Table Entry 

PSTN to ENTERPRISE_Voice:

Entry 1 (PSTN_TO_GV)

1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (PSTN_TO_GV) with priority 1.
3. Add the Destination Signaling Group which in this case is GOOGLE_SG.
4. In the Media panel, select DSP from the Audio Stream Mode and enable Media Transcoding.
5. Click OK.

Call Matching PSTN_TO_GV transformation table will be routed to the Google_SG.

Entry 2 (PSNT_TO_PBX)

1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (PSTN_TO_PBX) with priority 2.
3. Add the Destination Signaling Group which in this case is On-prem_PBX.
4. In the Media panel, select DSP from the Audio Stream Mode and enable Media Transcoding.
5. Click OK.

Call matching PSTN_TO_PBX transformation table will be routed to the On-prem_PBX SG.

GV to PBX_&_PSTN :

Entry 1 (GV_TO_PBX)

1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (GV_TO_PSTN) with priority 2.
3. Add the Destination Signaling Group (PRI_T1).
4. In the Media panel, select DSP from the Audio Stream Mode and enable Media Transcoding.
5. Click OK.

Call Matching GV_TO_PSTN transformation table will be routed to the PRI_T1 SG.

Entry 2 (GV_TO_PBX)

1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (GV_TO_PBX) with priority 1.
3. Add the Destination Signaling Group (On-prem_PBX).
4. In the Media panel, select DSP from the Audio Stream Mode and enable Media Transcoding.
5. Click OK.

Call Matching GV_TO_PBX transformation table will be routed to the On-prem_PBX SG.

PBX_TO_GV_&_PSTN:

Entry 1 (PBX_TO_PSTN)

1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (PBX_TO_PSTN) with priority 2.
3. Add the Destination Signaling Group (PRI_T1).
4. In the Media panel, select DSP from the Audio Stream Mode and enable Media Transcoding.
5. Click OK.

Call Matching PBX_TO_PSTN transformation table will be routed to the PRI_T1 SG.

Entry 2 (PBX_TO_GV)

1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (PBX_TO_GV) with priority 1.
3. Add the Destination Signaling Group (Google_SG).
4. In the Media panel, select DSP from the Audio Stream Mode and enable Media Transcoding.
5. Click OK.

Call Matching PBX_TO_GV transformation table will be routed to the Google_SG.

SBC Edge 1000 Configuration for PBX side

Media List - PBX

From the Settings tab, navigate to Media > Media List. Click the  icon at the top of the Media List View page.

1. Provide a name for the profile in the Description field.
2. Attach the Media Profiles by clicking Add/Edit.
3. Enable Dead Call Detection.
4. Click OK.

SIP Server Table - PBX

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. The Table Entries also contain links to counters that are useful for troubleshooting. The SIP Server supports either an FQDN or IP Address (V4 or V6).

From the Settings tab, navigate to SIP > SIP Server Tables. Click the  icon to create a new SIP Server Table.

1. Provide a name for the SIP Server in the Description field.
2. Click OK.

SIP Server Table Entry

1. Click on the SIP Server Table created in the previous step.
2. From the Create SIP Server drop-down menu, select IP/FQDN.
3. Provide IP Address and Port Number of the PBX.
4. Click OK.

SIP Signaling Group - PBX

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

From the Settings tab, navigate to Signaling Groups. Click Add SIP SG.

1. Attach the Call Routing Table (PBX_TO_GV_&_PSTN).
2. Attach the SIP Profile (Default SIP Profile).
3. Attach the SIP Server Table (on-prem_PBX).
4. Attach the Media List ID (on-prem_PBX).
5. Associate the appropriate IP address in the "Signaling/Media Source IP" field.
   1. This specifies the Logical IP address at which SIP messages are received.
   2. This address is used as the source IP for all SIP messages leaving the SBC 1000 through this Signaling Group.
6. Configure Protocol and Listen Ports in  the "Listen Ports" panel.
7. Create an entry in the Federated IP/FQDN panel.
   1. Federated IP addresses and FQDNs specified in a SIP Signaling Group are whitelisted. 
   2. The Federated IP/FQDN feature acts as an access control by defining from which server a SIP Signaling Group will accept messages.
8. Click OK.

SBC Edge 1000 Configuration for T1/PRI side

DSI Port Configuration

From the Monitoring tab, Select the DS1 port and make a configuration according to the service provider Trunk type, Framing, and Line coding.

SIP Signaling Group - PRI/PSTN

From the Settings tab, navigate to Signaling Groups. Click Add SIP SG.

1. Attach the Call Routing Table (PSTN_TO_ENTERPRISE_Voice).
2. Attach the Port Name (T1 Port 7:1).

SBC Edge 1000 Configuration for Google Voice SIP Link side

DNS

From the Settings tab, navigate to System > Node-Level Settings.

1. From the Use Primary DNS drop-down menu, select Yes.
2. Provide the Primary DNS IP address.
3. Select the Ethernet facing Google Voice SIP Link from the Primary Source drop-down menu.
4. Click Apply.

TLS Profile

TLS Profiles are used by SIP Signaling Groups when the TLS transport type is selected for incoming and outgoing SIP trunks (Listen Ports), and in SIP Server Tables when TLS is selected as the Server Host protocol.

From the Settings tab, navigate to Security > TLS Profiles. Click the  icon to create a new TLS profile.

1. From TLS Protocol drop-down menu, select TLS 1.0-1.2.
2. Add the cipher suites that are supported on Google Voice SIP Link (TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384 and TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256).
3. Disable the Validate Server and Client FQDN fields.
4. Click OK.

SDES-SRTP Profile

SDES-SRTP Profiles define a cryptographic context which is used in SRTP negotiation. SDES-SRTP Profiles are required for enabling media encryption and are applied to Media Lists.

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

1. Provide a name for the profile in the Description field.
2. Attach the Crypto suite "AES_CM_128_HMAC_SHAI_80", a crypto suite algorithm which uses the 128 bit AES-CM encryption key and a 80 bit HMAC_SHA1 message authentication tag length.
3. Set the Key Identifier Length to 0 to disable the MKI in SDP.
4. Click OK.

Media List - GV

From the Settings tab, navigate to Media > Media List. Click the  icon at the top of the Media List View page

1. Provide a name for the profile in the Description field.
2. Attach the Media Profiles by clicking Add/Edit.
3. Attach the SDES-SRTP profile (GV_BYOT).
4. Enable Dead Call Detection.
5. From the DTMF drop-down menu, select RFC2833.
6. Click OK.

Message Manipulation - GV

The Message Manipulation feature comprises two primary components that work in concert to modify SIP messages. These components are Condition Rules and Rule Tables. SIP Message rules per table include all SIP rule types: Header, Request, Status and Raw.

The Message Manipulation GOOGLE_RULE is used for the following purposes:

• To add the header “X-Google-Pbx-Trunk-Secret-Key” for Google Voice. The value of this header is generated when the SIP Trunk is created.
• To change the request URI of specific request messages to Google specified FQDN, trunk.sip.voice.google.com.

Message Rule Table

Message Rule can be added to: all messages, all requests, all responses or selected messages.

From the Settings tab, navigate to SIP > Message Manipulation > Message Rule Table. Click the  icon to create a Message Rule Table.

1. Provide a description for the Rule Table.
2. Apply Message Rule to the selected messages and choose Invite, Cancel, Options and ACK from the Message Selection list.
3. Click OK.

Message Rule Table Entry

Header Rule:

1. Click on the Message Rule Table GOOGLE_RULE.
2. From the Create Rule drop-down menu, select Header Rule.
3. Provide a name for the entry.
4. Add the header "X-Google-Pbx-Trunk-Secret-Key".
5. To add the value, select Add from the Header Value drop-down menu and provide the literal value of the header.
6. Click OK.

Request Line Rule:

1. Click on the Message Rule Table GOOGLE_RULE.
2. From the Create Rule drop-down menu, select Request Line Rule.
3. Provide a name for the entry in the Description field.
4. Replace the FQDN "siplink.telephony.goog" with "trunk.sip.voice.google.com" using regex.
5. Click OK.

SIP Profile - GV

From the Settings tab, navigate to SIP > SIP Profiles. Click the  icon to create a new SIP Profile.

1. Provide a name for the profile in the Description field.
2. Enable Session Timer.
3. Set the Minimum Acceptable Timer to 600 and the Offered Session Timer to 3600.
4. In the Options Tags panel, set the Timer field to Required and the Update field to Supported.
5. Click OK.

SIP Server Table - GV

From the Settings tab, navigate to SIP > SIP Server Tables. Click the  icon to create a new SIP Server Table.

1. Provide a name for the SIP Server in the Description field.
2. Click OK.

SIP Server Table Entry

1. Click on the SIP Server Table created in the previous step.
2. From the Create SIP Server drop-down menu, select IP/FQDN.
3. Provide the IP Address and the Port Number of the PSTN endpoint.
4. Enable OPTION pings by selecting SIP Options from the Monitor field.
5. Click OK.

SIP Signaling Group - GV

From the Settings tab, navigate to Signaling Groups. Click Add SIP SG.

1. Attach the Call Routing Table (GV_TO_PBX_&_PSTN).
2. Attach the SIP Profile (Google).
3. Attach the SIP Server Table (Google).
4. Attach the Media List ID (Google).
5. Associate the appropriate IP address in the "Signaling/Media Source IP" field.
6. Configure the Protocol, TLS Listen Ports and TLS Profile (Google_TLS) in the "Listen Ports" panel.
7. Provide the Google Voice SIP Link's FQDN or IP address in the Federated IP/FQDN panel.
8. Enable Message Manipulation and attach the profile Google_Rule to the Outbound Message Manipulation Table List.
9. Click OK.

Google Voice Configuration

For configuration on Google Voice, visit support.google.com/a?p=siplink.

Supplementary Services & Features Coverage

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

Legend

Caveats

The following items should be noted in relation to this Interop - these are either limitations, untested elements, or useful information pertaining to the Interoperability.

The below issues will be addressed by Google Voice in their upcoming releases.

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 & solutions, please visit:

https://ribboncommunications.com/products

Conclusion

This Interoperability Guide describes the successful configuration for Google Voice SIP Link interop involving the Ribbon SBC Edge 1000.

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 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.