---
title: WebRTC Voice Call Architecture
description: How to enable WebRTC Calls, Click2Call, WebCall from website or mobile app to Genesys Cloud.
---
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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.
### [](https://kb.trueengage.com/hubfs/TrueEngage%20Webrtc%20Architecture%20v2.png?hsLang=en) 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:
- [Genesys Cloud SIP Trunk Configuration](https://help.mypurecloud.com/articles/carrier-requirements-byoc-cloud/)
- [Firewall Configuration for Voice](https://kb.trueengage.com/genesys-cloud/firewall-configuration-for-webrtc-calls-end-user-networks-0?hsLang=en)
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