In this section:
The
This section outlines how to configure the
SBC Edge Connects to Zoom Phone BYOC Deployed in an Enterprise Network
To configure the
This best practice assumes that the
Installation Requirements
Product | Installation |
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Unable to show "metadata-from": No such page "_space_variables" | On KVM: Installing SBC SWe Lite on KVM Hypervisor On VMware ESXi: Installing SBC SWe Lite on VMware ESXi On Hyper-V: Installing SBC SWe Lite on Microsoft Hyper-V |
Unable to show "metadata-from": No such page "_space_variables" | |
Unable to show "metadata-from": No such page "_space_variables" |
TLS provides peer authentication, confidentiality and message integrity.
The prerequisites for TLS configuration between SBC Edge and Zoom are:
a Certificate Authority (CA)
Step | Action |
---|---|
1 | Generating a Certificate Signing Request |
2 | Importing a Trusted CA Certificate into SBC Edge |
3 | Verifying the Trusted Root CA Certificate |
4 | Configuring a TLS Profile |
Select Settings > Security > SBC Certificates > Generate SBC Edge CSR. The Generate Certificate Signing Request pane opens on the right.
Once the system generates the Certificate Signing Request (CSR), copy the result and send it to a Trusted Certificate Authority for their signature.
Once you receive the signed certificate from the Trusted Certificate Authority, perform Importing a Trusted CA Certificate into SBC Edge.
Importing a Trusted CA Certificate into SBC Edge
TLS Profile is required for the TLS handshake between SBC Edge and Zoom. This profile defines cipher suites supported by the SBC Edge. You must attach the TLS Profile to the SIP Server Table on the Zoom leg.
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. For more information on media profiles, refer to Managing Media Profiles.
SDES-SRTP Profiles define a cryptographic context which is used in SRTP negotiation. SDES-SRTP Profiles are required for enabling encryption and SRTP are applied to Media Lists.
Select Settings > Media > SDES-SRTP Profiles. The SDES-SRTP Profiles pane opens on the right.
This best practice does not detail the configuration on the Zoom Phone.
For more information on configuring SBC Edge to Zoom Phone BYOC, refer to "Section A: SBC Edge Configuration" in Ribbon SBC Edge Configuration with Zoom BYOC
The prerequisites for the Zoom Phone BYOC are:
an existing carrier that offers PSTN services or a router, gateway, or another SBC that supports SIP trunk connectivity
Make sure that your carrier provides Direct Inward Dialing (DID) numbers to use as external BYOC numbers.
a Zoom BYOC SIP trunk built between Ribbon SBC Edge and Zoom SBC
Make sure that the SBC Edge to Zoom BYOC trunk is over TLS/SRTP.
The following workflow defines the procedures for Configure SBC Edge for Zoom Phone BYOC.
SIP profiles control how the SBC Edge communicates with SIP devices. Configuring a SIP profile permits proper connectivity between the SBC Edge and the Zoom Phone BYOC service. For more information on SIP profiles, refer to Managing SIP Profiles.
Select Settings > SIP > SIP Profiles. The SIP Profile Table pane opens on the right.
Complete the following field:
• Description: Enter a name for the Zoom SIP profile entry (for example, Zoom profile).
Configuring a SIP profile permits proper connectivity between the SBC Edge and your enterprise IP PBX.
In the SIP Profile Table pane on the right, click Create SIP Profile. The Create SIP Profile Entry window opens.
Complete the following field:
• Description: Enter a name for the IP PBX SIP profile entry (for example, IP PBX Profile).
Media lists allow you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which must first be defined in Media Profiles. These lists allow you to accommodate specific transmission requirements, and SIP devices that only implement a subset of the available voice codecs. For more information on media lists, refer to Managing Media Lists.
SIP server tables contain information about the SIP devices connected to the
The entries in the SIP server tables provide information about the IP addresses, ports, and protocols used to communicate with each server. The entries also include links to counters that are useful for troubleshooting. For more information on SIP server entries, refer to Creating and Modifying Entries in SIP Server Tables.
Complete the following fields:
• Host FQDN/IP: Enter your Zoom IP address.
• Protocol: Select TLS.
• TLS Profile: Select Default TLS Profile.
• Monitor: Select SIP Options.
This configuration requires the IP address and port information to connect a SIP trunk to the
Start
Complete the following fields:
• Host FQDN/IP: Enter the IP address of your IP PBX.
• Protocol: Select UDP.
• Monitor: Select None.
Signaling groups allow the
Complete the following fields:
• Description: Enter a name for the Zoom Phone signaling group (for example, Zoom Phone_SG).
• Call Routing Table: Leave as Default Route Table for now.
You will create a call routing table for the Zoom Phone BYOC in Create a Call Routing Table and Entry from Zoom and will need to select the created call routing table for the Zoom phone BYOC in Change the Call Routing Table in the Zoom Phone Signaling Group Configuration.
• SIP Profile: Select the SIP profile you created in Configuring a SIP Profile for the Zoom Phone BYOC Service.
• SIP Server Table: Select the SIP server table you created in Configuring a SIP Server Table for the Zoom Phone BYOC Service.
• Media List ID: Select the media list you created in Configuring a Media List for the Zoom Phone BYOC Service.
• Signaling/Media Source IP: Select Ethernet 2 IP (10.0.0.0).
• Listen Port:
a. Click Create. The Create Listen Port window opens.
b. From the Protocol dropdown list, select TLS.
c. Click OK.
If you have created the TLS-5061 protocol before for other tasks, you will see the following error message when you click OK: "Validation Error - Unique item not unique for resource Listen Port, Protocol and TLS Profile ID configured is duplicate entry ID [error - parameter value missing]".
If you have created the TLS-5061 protocol before for other tasks, click Add/Edit to select TLS-5061.
• Federated IP/FQDN:
a. Click Create Federated IP/FQDN. The Add Federated IP/FQDN window opens.
b. In the IPv4/6 Address or FQDN field, enter your Zoom IP address.
c. Click OK.
Click OK.
Since your capacity may vary, 60 channels is an example value in the screen capture above.
Complete the following fields:
• Description: Enter a name for the enterprise IP PBX signaling group (for example, IP_PBX_SG).
• Call Routing Table: Leave as Default Route Table for now.
You will create a call routing table for the IP PBX in Create a Call Routing Table and Entry from the Carrier and will need to select the created call routing table for the IP PBX in Change the Call Routing Table in the IP PBX Signaling Group Configuration.
• SIP Profile: Select the SIP profile you created in Configuring a SIP Profile for an Enterprise IP PBX.
• SIP Server Table: Select the SIP server table you created in Configuring a SIP Server Table for the Enterprise IP PBX.
• Media List ID: Select the media list you created in Configuring a Media List for the Enterprise IP PBX.
• Signaling/Media Source IP: Select Ethernet1 IP (10.56.68.83).
• Listen Port:
a. Click Add/Edit. The Select Protocolss window opens.
b. From the Listen Port drop-down menu, select UDP-5060 and TCP-5060.
c. Click OK.
• Federated IP/FQDN:
a. Click Create Federated IP/FQDN. The Add Federated IP/FQDN window opens.
b. In the IPv4/6 Address or FQDN field, enter the IP address of your IP PBX.
c. Click OK.
Click OK.
Since your capacity may vary, 60 channels is an example value in the screen capture above.
Verify or create listening ports that the
Transformation tables facilitate the conversion of names, numbers, and other fields when routing a call. For example, transformation tables can convert a public PSTN number into a private extension number or SIP address (URI). Each call routing table entry requires a transformation table. Each Transformation Table contains a list of entries considered as routing rules to execute on. For more information on transformation tables and entries, refer to Managing Transformation Tables.
Complete the following fields:
• Description: Enter a name for the Zoom Phone BYOC transformation table entry.
• Match Type: Select the required match type (for example, Mandatory (Must Match)).
• Input Field Value: Enter the required value for the Called Address/Number (for example, (5126815704)).
• Output Field Value: Enter the required value for the Called Address/Number (for example, +1/1).
The configuration in the screen capture above passes the specified calling and called numbers from the IP PBX to Zoom and Zoom to IP PBX. The +1\1 value modifies the transformations (numbers) to add +1 to all calls sent to the Zoom Phone service and removes +1 from all calls sent to the carrier SIP trunk.
To pass all calling and called numbers, enter the .* value. For more information on regular expressions and transformation, refer to Regular Expressions for Number Matching and Transformation.
Click OK.
Call routing allows the
This section provides information on changing the default call routing tables you selected in Configuring a Signaling Group for the Zoom Phone BYOC Server and Configuring a Signaling Group for the Enterprise IP PBX.
Expand the signaling group you created for the Zoom Phone in Configuring a Signaling Group for the Zoom Phone BYOC Server.
From the Call Routing Table dropdown list, select the call routing table you created in Create a Call Routing Table and Entry from Zoom.
Use the following procedure to confirm that the
In the SBC Edge Real-Time Monitor pane, confirm that the channels under each signaling group you created in the Configure SBC Edge for Zoom Phone BYOC procedure are green. For details on channel status, refer to Monitoring Real Time Status.
Use the following procedure to place a test call.
Log into the WebUI (refer to Logging into the SBC Edge).
Select Diagnostics > Test a Call. The Test a Call pane opens on the right.
Complete the following fields:
• Destination Number: Enter the number assigned to a Zoom Phone user.
• Origination/Calling Number: Enter the number assigned to a local user.
• Call Routing Table: Select the call routing table that handles the call from the Zoom Phone BYOC.
See the following screen capture for an example configuration.
Click OK.