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Document Overview

This document provides the configuration details for Ribbon's SBC SWe Lite interworking with Twilio Elastic SIP Trunk, Microsoft Teams Direct Routing and Cisco Unified Communication Manager.

About Ribbon SBC SWe Lite

The Ribbon Session Border Controller Software Edition Lite (SBC SWe Lite) provides best-in class communications security. The SBC SWe Lite dramatically simplifies the deployment of robust communications security services for SIP Trunking, Direct Routing, and Cloud UC services. SBC SWe Lite operates natively in the Azure and AWS Cloud as well as on virtual machine platforms including Microsoft Hyper-V, VMware and Linux KVM.

About Twilio Elastic SIP Trunking

Twilio has developed an advanced SIP trunking service that addresses the key challenges that are holding back enterprises from realizing their communications transformation goals. Twilio Elastic SIP Trunking delivers global PSTN connectivity that enables enterprises to increase business agility, reduce costs and deliver uniform global reach.

About Microsoft Teams Direct Routing

Microsoft Phone System Direct Routing allows connection of a supported customer-provided Session Border Controller (SBC) to a Microsoft Phone System. Direct Routing enables using virtually any PSTN trunk with Microsoft Phone System and configuring interoperability between customer-owned telephony equipment, such as a third-party private branch exchange (PBX), analog devices, and Microsoft Phone System.

About Cisco Unified Communication Manager

Cisco Unified Communication Manager is a core call-control application of Cisco UCM. It provides enterprise-class call control, session management, voice, video, messaging, mobility and conferencing services in a way that is efficient, highly secure, scalable and reliable.

Scope

This document provides configuration best practices for deploying Ribbon's SBC SWe Lite with Cisco Unified Communication Manager (CUCM) and Microsoft Teams for Twilio Elastic SIP Trunking interop. 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.  

Non-Goals

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. 

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Ribbon SBC and the third-party product. Navigating the third-party product as well as the Ribbon SBC SWe Lite GUI is required. Understanding the basic concepts of TLS/TCP/UDP, IP/Routing, and SIP/SRTP is also necessary to complete the configuration and any required troubleshooting.

Prerequisites

The following aspects are required before proceeding with the interop:

• Amazon Web Services (AWS) subscription
• Ribbon SBC SWe Lite on AWS
• SBC SWe Lite License 
  • This interop requires the acquisition and application of cloud SIP sessions, as documented at Cloud-Based SBC SWe Lite Deployment Licenses
• Public IP Addresses
• Twilio Elastic SIP Trunk
  • Contact Twilio for Domain, IP and Port information
  • For more details, visit https://www.twilio.com/docs/sip-trunking or see the “Twilio “Twilio Elastic SIP Trunk Configuration” Configuration” section of this document
• TLS Certificates for SBC SWe Lite
  • Please refer to Working with Certificates

Product and Device Details

The configuration uses the following equipment and software:

Network Topology and E2E Flow Diagrams 

SBC SWe Lite - Twilio Deployment Topology 

SBC SWe Lite - Twilio Lab Topology 

Signaling and Media Flow

Document Workflow

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

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Installing SBC SWe Lite on AWS Installing SBC SWe Lite on AWS

Installing SBC SWe Lite on AWS

The SBC SWe Lite is available for deployment in AWS. It is created as a virtual machine (VM) hosted in AWS. To deploy an SBC SWe Lite instance, refer to Deploying an SBC SWe Lite via Amazon Web Services-AWS. Once SWe Lite instance is successfully created on AWS, retrieve the allocated NAT Public IPs, Ethernet IPs and Management IP. Also ensure ensure Twilio IP addresses are  are whitelisted on AWS access list. For more details, visit the link given in the References section.

SBC SWe Lite Configuration

Accessing SBC SWe Lite

Open any browser and enter the SBC SWe Lite IP address.

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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.

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For more details on Licenses, refer to Cloud-Based SBC SWe Lite Deployment Licenses.

Import Trusted Root CA Certificates

A Trusted CA Certificate is a certificate issued by a trusted certificate authority. Trusted CA Certificates are imported to the SBC SWe Lite to establish its authenticity on the network.

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

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View Networking Interfaces

The SBC SWe Lite supports five system created logical interfaces (known as Administrative IP, Ethernet 1 IP, Ethernet 2 IP, Ethernet 3 IP, and Ethernet 4 IP). In addition to the system created logical interfaces, the Ribbon SBC SWe supports user-created VLAN logical sub-interfaces.

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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).

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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.

Easy Configuration Wizard

Access the Easy Configuration Wizard

1. In the WebUI, click the Tasks tab.
2. In the left navigation pane, navigate to SBC Easy Setup > Easy Config Wizard. The Easy Configuration screen opens.

The SBC Edge WebUI provides a built-in Easy Configuration wizard that lets you quickly and easily deploy the SBC for operation with provider endpoints (SIP trunk, ISDN PSTN trunk, or IP PBX trunk) and user endpoints (Microsoft Teams, Microsoft On Premises - Skype for Business/Lync, IP Phones, or ISDN PBX or IP PBX).

Navigating the Wizard

As the wizard runs, it directs you through three configuration steps:

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• Previous: Moves back to the previous step.
• Next: Advances to the next step when the current step is validated and complete.
• Finish: Submits the data to the SBC.
• Cancel: Cancels the Easy Configuration data entered and redirects to the main WebUI.

Configure SBC SWe Lite using Easy Configuration Wizard

During this interop:

• Multi-legged approach was used to configure Twilio US SIP Trunk and Microsoft Teams (Application: SIP Trunk ↔ Microsoft Teams)
• Single-legged approach was used to configure Twilio EMEA SIP Trunk (Application: SIP Trunk) 
• Single-legged approach was used to configure CUCM (Application: IP PBX)

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Configure SBC SWe Lite for Twilio US Trunk and for Microsoft Teams

Step 1: Configure US Trunk for Twilio along with Microsoft Teams using Multi-legged approach by following the steps below:

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• A pop up window appears once all the 3 steps are completed. Click OK to continue.
• Wait for the configuration to complete and click OK on the next window. This will complete the configuration of Twilio US Trunk and Microsoft Teams.

Configure SBC SWe Lite for Twilio EMEA Trunk

Step 1: Use Single-legged approach for Twilio EMEA Trunk configuration.

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• A pop up window appears once all the 3 steps are completed. Click OK to continue.
• Wait for the configuration to complete and click OK on the next window. This will complete the configuration of Twilio EMEA Trunk.

Configure SBC SWe Lite for CUCM

Step 1: Use the Single-legged approach to configure IP PBX.

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• A pop up window appears once all the 3 steps are completed. Click OK to continue.
• Wait for the configuration to complete and click OK on the next window. This will complete the configuration of CUCM leg on SBC SWe Lite.

Modify SBC SWe Lite Configuration 

The Easy Configuration Wizard does not currently set all Twilio applicable variables to the correct settings. This will be addressed in the subsequent SBC SWe Lite releases. Until then, please follow the procedures below.

Assign NAT Public IP

Change the settings on all the SGs as follows:

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Enable Static NAT and map the respective IP addresses.

Enable OPTIONS

An OPTIONS message is sent to the server. When this option is selected, additional configuration items are displayed:

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Modify SIP Profiles

Enable Session Timers

From the Settings tab, navigate to SIP > SIP Profiles, Enable Session Timers and set the Timer as Required on all the SIP Profiles.

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• Send Assert Header - Never- When disabled, privacy information in the outbound INVITE is sent depending on the configuration of the Trusted Interface and the Privacy Pass-through Header.
• Trusted Interface - Disable.

Enable Dead Call Detection

Specifies whether or not to use RTCP-based Dead Call Detection (DCD).

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• Enable DCD from the options provided in the drop-down.

SBC SWe Lite Configuration for Twilio TLS/SRTP Trunk (Recommended)

This section describes the steps to configure SBC SWe Lite with TLS/SRTP towards Twilio SIP Trunk. Ribbon strongly recommends encrypting the connection between Twilio SIP Trunk and SBC SWe Lite.

Create SRTP Profile

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

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1. Provide the desired description for the profile.
2. Set Operation Option as "Required". This setting permits call connections only if you can use encryption for the call. If the peer device does not support SRTP (Secure Real Time Protocol) for voice encryption over the IP network, the call setup will fail.
3. 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.
4. Key Identifier Length set to "0" - Set this value to 0 to disable the MKI in SDP.
5. Click OK.

Attach SRTP Profile to the Media List

From the Settings tab, navigate to Media > Media List, Click the expand () icon next to the entry.

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1. Attach the SDES-SRTP profile (Specifies the profile for authentication/encryption protocols applied with this Media List) created in the previous step.
2. Click Apply.

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Update Signaling Group

Signaling Groups allow grouping telephony channels 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.

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1. Update the Federated IP/FQDN(Only if the FQDNs for TLS are different)..
2. Click the  icon to add Listen Ports for TLS. 
3. Use TLS as the Protocol and update the Port Number provided by the Service Provider (Port Number 5061 was used during this interop).
4. Click Apply.

Update SIP Server Table

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.

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Configure Transformation Tables

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

From the Settings tab, navigate to Transformation.

To Modify a Transformation Table

The Transformation Tables are created for MS Teams and Twilio US Trunk (TEAMS-TWILIO_US: From Microsoft Teams Direct Routing: Passthrough and TEAMS-TWILIO_US: From SIP Trunk: Passthrough respectively) through Easy Config Wizard. These are modified to allow specific patterns to reach the destination Signaling Group. 

1. Click the expand () icon next to the entry you wish to modify.
2. Modify the table's Description as desired.
3. Modify the Values from Input field and Output field as required.
4. Set the Match Type as Optional (Match one).
5. Click OK.

To Create a Transformation Table

Each Transformation Table contains a list of entries considered as routing rules to execute on. Each rule is executed in order until the end of the table is reached or when a Mandatory entry fails to execute.

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Follow the same procedure to create Transformation Tables for CUCM.

Creating an Entry to a Message Transformation Table

For this interop, the entries are created based on the numbers associated with each endpoint. Users are free to select their own variables or Regular expressions.

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Configure Call Routing Tables

Call Routing allows carrying of calls between Signaling Groups. Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried, and how they are translated. 

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The Call Routing Tables are created to route the calls between TEAMS-TWILIO_US: Teams Direct Routing SG and TEAMS-TWILIO_US: Border Element SG through Easy Config Wizard. The user is allowed to modify these tables as per the requirement.

Modifying an Entry to a Call Routing Table

1. Click the expand () icon next to the entry you wish to modify.
2. Edit the entry properties as required.

Creating an Entry to a Call Routing Table

Call Routing 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).

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1. Click the Create Routing Entry ( ) icon.

2. Set the following fields:

   Admin State: 
   Enabled - Enables the call route entry for routing the call, displays in configuration header as .

   Route Priority: 
   Priority of the route from 1 (highest) to 10 (lowest). Higher priority routes are matched against before lower priority routes regardless of the order of the routes in the table.

   Number/Name Transformation Table:
   Specifies the Transformation Table to use for this routing entry. This drop down list is populated from the entries in the Transformation Table. 

   Destination Signaling Groups:
   Specifies the Signaling Groups used as the destination of calls. The first operational Signaling Group from the list is chosen to place the call. Click the Add/Edit button to select the destination signaling group.

   Audio Stream Mode:
   DSP (default entry): The SBC uses DSP resources for media handling (transcoding) but it does not facilitate the capabilities/features between endpoints that are not supported within the SBC (codec/capability mismatch). When DSP is configured, the Signaling Groups enabled to support DSP are attempted in order.

   Media Transcoding:
   Enabled: Enable Transcoding on SIP-to-SIP calls.

3. Click Apply.

Creating Multiple Entries to a Call Routing Table

SBC SWe Lite allows the user to create multiple entries to a Call Routing table. As there are four SIP Signaling Groups in this deployment, it is required to create multiple route entries to allow the call to reach a specific destination SIP Signaling Group.

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Message Manipulation 

All the calls initiated from Teams endpoint will have "PRIVACY: id" header. As Trusted interface is disabled on Twilio (US and EMEA) SIP profiles, SWe Lite sends out all the calls as Anonymous. In order to avoid this, we have used an SMM on the Inbound Message Manipulation list of TEAMS-TWILIO_US: Teams Direct Routing SIP SG. 

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The Message Manipulation feature comprises two primary components that work in concert to modify SIP messages. Those component are Condition Rules and Rule Tables.

Creating a Condition Rule Table

Condition rules are simple rules that apply to a specific component of a message (e.g., diversion.uri.host, from.uri.host, etc.) the value of the field specified in the Match Type list box can match against a; literal value, token, or REGEX.

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• Provide a suitable description for the rule.
• From the Match type drop-down, select "from" as we are checking if the From header has Anonymous or calling party number.
  Match type specifies the first operand for the logical condition expressed by this rule. The operand must be a parameter tree token identifier.

• Use Regex Operation.
  Operation specifies the match type for this condition.

• Write a Regular Expression to match everything but Anonymous.
• Click OK.

Creating a SIP Message Rule Table

From the Settings tab, navigate to SIP > Message Manipulation > Message Rule Table. Click the Create Message Rule Table() icon.

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• Header Action: Remove (if the header is present, it is dropped from the message).
• Header Name: Specifies the type of header referenced by this rule. In this case, Privacy header.
• Click Apply.

Attaching the Message Table to SIP SG

From the Settings tab, navigate to Signaling Groups > TEAMS-TWILIO_US: Teams Direct Routing.

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• This displays a drop-down list of available message tables. Select an entry and click Apply.

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Twilio Elastic SIP Trunk Configuration Twilio Elastic SIP Trunk Configuration

Twilio Elastic SIP Trunk Configuration

From your Twilio Console, navigate to the Elastic SIP Trunking area (or click on the  icon on the left vertical navigation bar).

Create an IP-ACL rule

Click on Authentication in the left navigation, and  then click on IP Access Control Lists. 

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Create a new IP-ACL, for example call it "Ribbon" and add your SBCs IP addresses (Kindly refer to the section Ribbon Installing SBC Edge SWe Lite R9.0 on AWS Interop with Cisco UCM and Microsoft Teams Direct Routing for Twilio Elastic SIP Trunking : Interoperability Guide)

Create a new Trunk

For each geographical region desired (eg. North America, Europe), create a new Elastic SIP Trunk.

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Note: Here is where you can enable the use of TLS & SRTP on your Trunk, learn more here.

In the Termination section, select a Termination SIP URI. 

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Note: If you enabled “Secure Trunking”, then you need to include the “transport=tls” parameter in your Origination URIs, learn more here.

Continue to add the other Origination URIs, so you have the following configuration:

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In this example, Origination traffic is first routed via Twilio’s Ashburn edge, if that fails then we'll route from Twilio’s Umatilla edge.

Associate your Twilio Phone Numbers on your Trunk

In the Numbers section of your Trunk, add the Phone Numbers that you want to associate with each Trunk. Remember to associate the Numbers from a given country in the right Trunk. For example, associate US & Canada Numbers with the North American Trunk and European Numbers with the European Trunk etc.

CUCM Configuration

Accessing CUCM (Cisco Unified CM Administration)

1. Open browse and enter the CUCM IP Address.
2. Select Cisco Unified CM Administration from the Navigation drop-down.
3. Provide the credentials and click Login.

Configure SIP Trunk Security Profile

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.

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• Provide the desired Name and Description.
• Choose Non Secure from Device Security Mode.
      - No security features except image authentication apply. A TCP or UDP connection opens to Unified Communications Manager.
• From Incoming Transport Type, select TCP+UDP.
      - When Device Security Mode is Non Secure, TCP+UDP specifies the transport type.
• Select Outgoing Transport Type as UDP.
• Click Save.

Configure SIP Profiles

A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that you cannot delete or change.

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• Enable SIP OPTIONS Ping.
      - SIP OPTIONS are requests to the configured destination address on the SIP trunk.
• Click Save.

Configure Media Resource Group

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.

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• Enter a unique name in this required field to identify the media resource group.
• Enter a description for the media resource group. 
• To add a media resource for this media resource group, choose one (MoH_2 in this case) from the available Media Resources list and click the down arrow. After a media resource is added, its name moves to the Selected Media Resources pane.

• Click Save.

Configure Media Resource Group List

A Media Resource Group List provides a prioritized grouping of media resource groups. An application selects the required media resource, such as a music on hold server, from among the available media resources according to the priority order that is defined in a Media Resource Group List.

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• Enter a unique name in this required field to identify the Media Resource Group List.

• Choose the Media Resource Group created in the previous step from the Available Media Resource Groups list and click the down arrow that is located between the two panes. After a media resource group is added, its name moves to the Selected Media Resource Groups pane.

• Click Save.

Trunk Configuration

Use a trunk device to configure a logical route to a SIP network.

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Configure Call Routing

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.

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• This way of configuring Route Pattern requires additional settings to remove the digits before the Dot.
• From Discard Digits drop-down, choose PreDot.
      - This would remove the digits which are present before the Dot (1 in this case).

Configure End Users

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 you associate a phone in the End User Configuration window. 

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• Enter the unique end user identification name.
• Enter alphanumeric or special characters for the end user password and confirm the same.
• Enter numeric characters for the end user PIN and confirm.
• Enter the end user last name.
• For Digest Credentials, enter a string of alphanumeric characters and confirm.

Phone Setup

• In Cisco Unified Communications Manager Administration, use the Device > Phone menu path to configure phones.
• Click Add New.

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• Click Apply Config followed by the Reset button.
• Reset, Restart and Close the window.

Device Association

• Navigate back to User Management > End User.
• In the Device Information field, click Device Association. This will display all the available devices. 

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• After selecting the appropriate device, it will appear in the Controlled Devices pane.

Enable MoH

In Cisco Unified Communications Manager Administration, use the System > Service Parameters menu path to configure service parameters.

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• Set the Duplex Streaming Enabled flag to True. This parameter determines whether Music On Hold (MOH) and Annunciator use duplex streaming.
• Click Save.

Configuration for SIP-URI calling

The SIP URI scheme is a Uniform Resource Identifier(URI) scheme for the Session Initiation Protocol(SIP) multimedia communications protocol. 

Configure End user

• In Cisco Unified CM Administration, navigate to User Management > End User.
• Click Find. This will display all the end users created.

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• Provide a SIP address in user@domain.tld format.
• Click Save.

Configure Route

Cisco Unified Communications Manager uses SIP route patterns to route or block both internal and external calls.

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• For Domain Routing pattern usage, enter a domain name(interopdmain.com in this case) IPv4 Pattern field that can resolve to an IPv4 address.
• From the drop-down list choose the SIP trunk created earlier to associate the route pattern.
• Click Save.

Directory Number Information

Using Cisco Unified Communications Manager Administration, you configure and modify directory numbers (DNs) that are assigned to specific phones. 

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• Click on the Directory number that needs a Directory URI assigned.
• Add the SIP-URI and save.
• Click Apply Config, Reset and Restart for the configuration to reflect.

MS TEAMS Configuration

For Microsoft Teams Direct Routing configuration for SBC SWe Lite, refer to the following: Connect SBC Edge to Microsoft Teams Direct Routing

Please check the connectivity for interfacing with Microsoft Teams Direct Routing before making the calls by following the procedure provided at the following link: Working with Connectivity Check - Verifying Service and Port Requirements for CCE and Teams

Monitor Real Time Status

Place a Test Call

Access SBC SWe Lite's WebUI and click the Monitor tab. Confirm all the SIP Signaling Groups are active. This panel provides current information on the status of Ports, Channels and in-progress Calls on the Ribbon SBC SWe Lite system.

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• Place a test call from Microsoft Teams client to PSTN.
• Make sure the PSTN is presented with an incoming call(Phone display).
• TEAMS-TWILIO_US: Teams Direct Routing SIP Signaling Group and TEAMS-TWILIO_EMEA: Border Element SIP Signaling Group present an alerting indication (magenta) in the respective channels. Click on the seized channels for the details.

Answer Call and Confirm Connection

• Answer the call on PSTN endpoint.
• TEAMS-TWILIO_US: Teams Direct Routing SIP Signaling Group and TEAMS-TWILIO_EMEA: Border Element SIP Signaling Group present a connected indication (blue) in the respective channels. Click on the seized channels for the details.

Disconnect the Call

• Disconnect the call and ensure that the Channel Status is Idle.

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Supplementary Services and Features Coverage

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

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Legend

Caveats

Note the following items in relation to this Interop:

• OPUS codec with Asymmetric Payload negotiation is not supported. Hence, Customers are recommended to use Symmetric Payload type on both the ends.
• MS Teams does not support SIP-URI calling with Direct Routing. The SIP-URI testing has been done only from CUCM to MS Teams via SBC SWe Lite.

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/about-us

References

For detailed information about Ribbon products and solutions, please visit:
https://ribboncommunications.com/products

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For additional information on Ribbon SBC SWe Lite on AWS, please visit:
Deploying an SBC SWe Lite via Amazon Web Services-AWS

For additional information on Teams, please visit:

Best Practice - Troubleshoot Issues with Microsoft Teams Direct Routing and Connect SBC Edge to Microsoft Teams Direct Routing

For detailed information about Twilio Elastic SIP Trunking and solutions, please visit:
https://www.twilio.com/sip-trunking, https://www.twilio.com/docs/sip-trunking and https://www.twilio.com/docs/sip-trunking/elastic-sip-trunking-solution-blueprints

Conclusion

This Interoperability Guide describes successful configuration for Twilio Elastic SIP Trunking interop involving Ribbon SBC SWe Lite on AWS, Cisco Unified Communication Manager and Microsoft Teams Direct Routing.

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