TrueEngage and Genesys Cloud Integration for WebRTC Calls

TrueEngage integrates with Genesys Cloud to facilitate seamless WebRTC calls through a SIP trunk, leveraging Telnyx as the WebRTC provider. This integration enables users to initiate voice calls directly from the TrueEngage widget on their website, which are then routed to Genesys Cloud for further handling.

How the Integration Works

1. Visitor Initiates WebRTC Call:
   • A website visitor initiates a call from the TrueEngage widget.
   • The call is routed through WebRTC to a dedicated DID (Direct Inward Dialing) number.
   • Telnyx provides the DID and acts as the WebRTC infrastructure provider, ensuring reliable voice communication.
2. Processing the Call with Telnyx:
   • The WebRTC call is routed to Telnyx, where it is converted into a SIP call.
   • SIP Trunking: TrueEngage automatically configures a SIP trunk during installation, which connects Telnyx and Genesys Cloud. The call is pushed to the SIP trunk for further processing.
3. Inbound Call Routing to Genesys Cloud:
   • Genesys Cloud receives the call through the SIP trunk, with routing determined by the DID number assigned to the call.
   • The Genesys Cloud administrator assigns the appropriate Call Flow to the DID, ensuring that the call is directed to the right queue or agent.
4. Call Handling in Genesys Cloud:
   • Once the call reaches Genesys Cloud, the assigned Call Flow processes it by either placing the caller in a queue or routing them to a specified agent.
   • The administrator configures Genesys to ensure the correct routing and handling of the call.

Technical Configuration

• SIP Trunk Setup:
  • The SIP trunk is set up in Genesys Cloud during the installation of the TrueEngage widget.
  • The trunk uses FQDN routing provided by Telnyx and is configured with TLS on port 5061 for secure communication.
• DID Configuration:
  • The DID numbers are provisioned by Telnyx and are automatically configured in Genesys Cloud when setting up the widget.
  • Each DID has a specific Call Flow assigned by the Genesys administrator.
• Call Flow Assignment:
  • It's critical for the Genesys administrator to assign the correct Call Flow to each DID number to ensure calls are routed correctly.

Security & Network Requirements

• TLS and Media Stream Security:

  • SIP communication between Telnyx and Genesys Cloud is secured using TLS on port 5061.
  • RTP media streams are handled via Telnyx Media Servers, and specific IP addresses need to be whitelisted to ensure proper communication.

• Required IP Addresses:

  • The following Telnyx IP addresses must be whitelisted for secure communication and media handling:

  Sip Trunk: - 185.246.41.140 - 185.246.41.141

    **Media:**
    
    - 50.114.136.128/25
    - 50.114.144.0/21
    - 64.16.226.0/24
    - 64.16.227.0/24
    - 64.16.228.0/24
    - 64.16.229.0/24
    - 64.16.230.0/24
    - 64.16.248.0/24
    - 64.16.249.0/24
    - 103.115.244.128/25
    - 185.246.41.128/25
  

Additional Information

For more detailed setup and configuration instructions for your Genesys Cloud and TrueEngage integration, please refer to the relevant documentation:

• Genesys Cloud SIP Trunk Configuration
• Telnyx SIP Info