Introduction

This is a web component that allows users to initiate a SIP call directly from their web browser using WebRTC technology. The component provides a simple user interface that includes a phone number input field and a "Call" button. When the user clicks the "Call" button, the component establishes a SIP connection using the specified phone number and initiates a WebRTC session that enables real-time audio communication.

Installation

To use this web component in your project, you can either download the source code and include it in your project manually, or install it using a package manager like npm:

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Usage

To use the Click-to-Call SIP WebRTC Component in your web application, you can simply import the component and include it in your HTML markup:

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You can also customize the component by passing props to it. For example, you can specify the SIP server URL and credentials, the audio and video settings, and the UI elements to display:

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For more information on the available props and events, please refer to the API documentation.

API

The Click-to-Call SIP WebRTC Component exposes the following props and events:

Props

· sip-server: The URL of the SIP server to connect to (default: undefined)

· sip-username: The username to use for SIP authentication (default: undefined)

· sip-password: The password to use for SIP authentication (default: undefined)

· audio-enabled: Whether to enable audio communication (default: true)

· video-enabled: Whether to enable video communication (default: false)

· display-input: Whether to display the phone number input field (default: true)

· display-button: Whether to display the "Call" button (default: true)

Events

· call-started : Fired when the user initiates a call.

· call-ended : Fired when the call ends or is terminated by the user.

License

This component is released under the MIT license. See the LICENSE file for more details.