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• Ribbon SBC SWe Edge
• Ribbon SBC SWe Edge license
  
  • This interop requires the acquisition and application of SIP sessions, as documented at Working with Licenses
• Public IP addresses
• TLS certificates for SBC SWe Edge
  • 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

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To deploy Ribbon SBC SWe Edge instance, refer to Installing SBC SWe Edge.

Ribbon SBC SWe Edge Configuration

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For more details on Licenses, refer to Working with Licenses.refer to SWe Edge License

SBC Certificate

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

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• For TLS to work, a Trusted CA (Certificate Authority) is required. For this interop, GoDaddy is used as Trusted CA.
• Add an entry in the Public DNS to resolve Ribbon SBC SWe Edge FQDN to Public IP Address.
• Ensure to have the following certificates as part of the root certificate trust:
  • GTS Root R1
  • GlobalSign R2 Root CA (if required)

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Follow the steps above to import GTS Root R1 and GlobalSign R2 Root CA certificates from Google Voice.

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Networking Interfaces

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

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

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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 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.

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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 (PSTN_TO_GV).
2. Attach the SIP Profile (PSTN_SIP_PROFILE).
3. Attach the SIP Server Table (PSTN).
4. Attach the Media List ID (PSTN).
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 SWe Edge or SBC 1000/2000 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. Enable Message Manipulation and attach the profile PSTN_RULE to Inbound Message Manipulation Table List.
   1. This option allows the SBC to manipulate SIP messages using previously configured Message Tables.
9. Click OK.

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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.

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1. Provide a name for the profile.
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.

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• 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.
• To modify the FQDN in the To header to trunk.sip.voice.google.com.

Message Rule Table

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

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

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. Select Header Action as Modify and choose To from the Header Name list.
5. Replace the FQDN "siplink.telephony.goog" with "trunk.sip.voice.google.com" using regex.
6. Click OK.

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SIP Profile - GV SIP Profile - GV SIP Profile - GV

SIP Profile - GV

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

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1. Attach the Call Routing Table (GV_TO_PSTN).
2. Attach the SIP Profile (GOOGLE_SIP_PROFILE).
3. Attach the SIP Server Table (GOOGLE).
4. Attach the Media List ID (GOOGLE).
5. Select the SDES-SRTP Profile GV_BYOT in the Proxy Local SRTP Crypto Profile ID field.
6. Associate the appropriate IP address in the "Signaling/Media Source IP" field.
7. Configure the Protocol, TLS Listen Ports and TLS Profile (GOOGLE_TLS) in the "Listen Ports" panel.
8. Provide the Google Voice SIP Link's FQDN or IP address in the Federated IP/FQDN panel.
9. Enable Message Manipulation and attach the profile GOOGLE_RULE to the Outbound Message Manipulation Table List.
10. Click OK.

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Call Routing Table Entry

Call Routing entries must to be created after creating SIP Signaling Groups as Destination SGs need to be attached to these entries.

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1. Click the Create Routing Entry ( ) icon.
2. Attach the Transformation Table (PASSTHROUGH_GV).
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.

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

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

Supplementary Services & Features Coverage

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• Short Code calls are not supported on Google Voice clients.
• When GV rejects or does not send CANCEL request when it terminates answer the call before the far-end answers.GV does not send ACK for 4xx and 5xx error responses.from PSTN, the call is expected to connect to GV Voice Mail after 30 seconds. However, the SWe Edge sends a CANCEL to GV to terminate the call before it connects. 

These issues will be addressed by GV/Ribbon in their upcoming releases.

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

References

For detailed information about Ribbon products & solutions, please visit:

https://ribboncommunications.com/products

Conclusion

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

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