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WebRTC Voice Call  Architecture

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 website widget or a mobile app deployment.

WebRTC_diagram

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:

    • 36.255.198.128/25
    • 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: