voipstack_audio_fork

SIP proxy that forks RTP audio streams to multiple outputs (files, WebSocket, etc).

Features

• Acts as a SIP proxy between PBX and endpoints
• Captures and dumps RTP audio streams
• Support for PCMU codec
• Configurable output destinations

Installation

From GitHub Releases

wget https://github.com/voipstack/voipstack_audio_fork/releases/latest/download/voipstack_audio_fork_linux_x86_64.tar.gz
tar -xzf voipstack_audio_fork_linux_x86_64.tar.gz
sudo mv voipstack_audio_fork /usr/local/bin/
sudo chmod +x /usr/local/bin/voipstack_audio_fork

From Source

shards install
shards build
cp bin/voipstack_audio_fork /usr/local/bin/voipstack_audio_fork

Production

For production deployment use systemd service unit. The file should be located at /etc/systemd/system/voipstack_audio_fork.service.

voipstack_audio_fork.service

wget https://github.com/voipstack/voipstack_audio_fork/releases/latest/download/voipstack_audio_fork_linux_x86_64.tar.gz
tar -xzf voipstack_audio_fork_linux_x86_64.tar.gz
sudo mv voipstack_audio_fork /usr/local/bin/
sudo chmod +x /usr/local/bin/voipstack_audio_fork

wget https://raw.githubusercontent.com/voipstack/voipstack_audio_fork/refs/heads/main/voipstack_audio_fork.service
sudo cp voipstack_audio_fork.service /etc/systemd/system/voipstack_audio_fork.service

sudo useradd voipstack_agent
sudo systemctl enable voipstack_audio_fork
sudo systemctl start voipstack_audio_fork

Usage

Basic Usage

./bin/voipstack_audio_fork -s sip://pbx_host:5080 -l 0.0.0.0 -p 5060 -o raw:///tmp/audio.ulaw

CLI Options

• -s, --pbx=URL - PBX URL (e.g., sip://192.168.1.100:5080)
• -l, --listen=HOST - Listen host (default: 127.0.0.1)
• -p, --port=PORT - Listen port (default: 5060)
• -o, --output FILE - Output destination (format: raw://path)
• -h, --help - Display help message

Example Setup

Forward SIP traffic from port 5060 to PBX at 192.168.1.10:5080 and dump audio:

./bin/voipstack_audio_fork \
  -s sip://192.168.1.10:5080 \
  -l 172.15.238.1 \
  -p 5060 \
  -o raw:///var/audio/call.ulaw

Freeswitch Usage

Test with playback.

freeswitch@096f2ec85864> originate sofia/internal/voipstack@172.15.238.1:5060 &playback(/etc/freeswitch/audios/stones-karaoke.wav)

Library Usage

require "voipstack_audio_fork"

class CustomDumper < VoipstackAudioFork::MediaDumper
  def start(session_id : String)
    # Initialize session
  end

  def dump(session_id : String, data : Bytes)
    # Process RTP data
  end

  def stop(session_id : String)
    # Cleanup session
  end
end

server = VoipstackAudioFork::Server.new
server.bind_pair("0.0.0.0", 5060, "pbx.example.com", 5080)
server.attach_dumper(CustomDumper.new)
server.listen

Architecture

• VoipstackAudioFork::Server - Main SIP proxy server
• VoipstackAudioFork::MediaDumper - Abstract base for audio processors
• Handles SIP INVITE, ACK, and BYE methods
• Dynamically spawns UDP servers for each RTP session

Caveats

• Only supports PCMU codec
• Single call recording per session

Development

crystal spec
crystal tool format

Contributing

1. Fork it (https://github.com/bit4bit/voipstack_audio_fork/fork)
2. Create your feature branch (git checkout -b my-new-feature)
3. Commit your changes (git commit -am 'Add some feature')
4. Push to the branch (git push origin my-new-feature)
5. Create a new Pull Request

License

MIT

Contributors

• Jovany Leandro G.C - creator and maintainer