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WebRTC Voice Call  Architecture

TrueEngage integrates with Genesys Cloud to facilitate seamless WebRTC calls through a SIP trunk, leveraging Vonage as the WebRTC provider. This integration enables users to initiate voice calls directly from the TrueEngage website widget, kiosk or mobile app deployment.

audio-data-flow-vonage.drawio

How the Integration Works

  1. Visitor Initiates WebRTC Call:
    • A website visitor initiates a call from the TrueEngage widget.
    • TrueEngage uses Vonage API to create the WebRTC session
  2. Processing the Call with Vonage:
    • The WebRTC call is routed to Vonage, where it is converted into a SIP call.
    • SIP Trunking: TrueEngage automatically configures a SIP trunk during installation, which connects Vonage and Genesys Cloud. The call is forwarded 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 Vonage and is configured with TLS on port 5061 for secure communication.
  • DID Configuration:
    • The DID numbers are provisioned by TrueEngage 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 Vonage and Genesys Cloud is secured using TLS on port 5061.
    • RTP media streams are handled via Vonage Media Servers

Additional Information

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