baresip（baresip pcm）

Baresip is an open-source, portable, and modular SIP user agent. It is designed to provide a simple and flexible platform for voice and video communication over IP networks. In this article, we will explore the various features and functionalities of Baresip.

## Installation

To install Baresip, follow these steps:

1. Download the latest version of Baresip from the official website.

2. Extract the downloaded file to a desired location on your computer.

3. Open a terminal and navigate to the extracted directory.

4. Run the `make` command to compile and build Baresip.

5. Once the installation process is complete, you can run Baresip using the `./baresip` command.

## Configuration

Before using Baresip, you need to configure it according to your preferences and network settings. The configuration file can be found at `~/.baresip/config`. Open this file in a text editor and make the necessary changes.

The configuration file allows you to set various parameters such as SIP server address, audio and video codecs, sound device settings, and more. It is recommended to consult the Baresip documentation or the official website for a detailed explanation of the available configuration options.

## Usage

Once Baresip is installed and configured, you can start using it for voice and video communication. Baresip provides a command-line interface (CLI) which allows you to interact with the user agent. Here are some basic commands to get started:

- **dial**: Use the dial command followed by the SIP address or phone number to make a call.

- **answer**: Use the answer command to answer an incoming call.

- **hangup**: Use the hangup command to end an ongoing call.

- **mute**: Use the mute command to mute your microphone during a call.

- **video**: Use the video command to enable or disable the video stream during a call.

You can also use the Baresip GUI (Graphical User Interface) to interact with the user agent. The GUI provides a more user-friendly interface with additional features such as an address book, call history, and settings menu.

## Extensions and Plugins

One of the key advantages of Baresip is its modular architecture. It allows users to extend its functionality by adding custom modules and plugins. The Baresip community has developed a wide range of extensions and plugins that can be easily integrated into the user agent.

Some popular extensions and plugins include:

- **STUN/TURN**: Adds support for STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) protocols.

- **SRTP**: Adds support for Secure Real-time Transport Protocol for encrypted voice and video communication.

- **ICE**: Adds support for Interactive Connectivity Establishment protocol for NAT traversal.

- **Codec plugins**: Adds support for additional audio and video codecs.

You can find a comprehensive list of available extensions and plugins on the official Baresip website or in the Baresip community forum.

## Conclusion

Baresip is a versatile and customizable SIP user agent that provides a wide range of features for voice and video communication. Its portable and modular architecture makes it suitable for various platforms and network environments. Whether you are a casual user or a developer, Baresip offers a flexible solution for your communication needs.