Capstone Project for Georgia Tech

Releases

Implements liblinphone to create a SIP endpoint that does some pretty specific things.

It is meant to combine the low bandwidth of a voice codec for collaboration on projects that require "studio quality" outputs. The "studio quality" is 320kpbs MP3, though this could easily be altered with little issue.

It should work with all SIP servers, though it was built and tested with FreePBX/Asterisk.

• User status synchronization (recording, mic disabled, speakers disabled)
• Records locally the entirety of the session in voice quality
• Allows 'artists' to record locally parts or all of the session in studio quality
• Allows 'artists' to send these studio quality recordings at the end of the session
• Allows 'producers' to segment audio (studio and voice quality) into arbitrary chunks in real time
• Allows playback of these chunks in voice quality
• Allows for an arbitrary number of users, limited only by SIP server settings and accounts.

Python 2.7

Precompiled binaries coming later.

linphone

• [Windows] pip install linphone
• [OS X] pip install
  • Download linphone.whl
  • pip install linphone.whl
  • We could not get a newer version to install on 10.10.

Kivy

• [Windows] https://kivy.org/docs/installation/installation-windows.html
• [OS X] https://kivy.org/docs/installation/installation-osx.html#using-homebrew-with-pip

PyDub and PyAudio

• Requires ffmpeg or libav (we are using ffmpeg)
• Windows
  • Install ffmpeg and add it to path
  • pip install pyaudio pydub
• OSX
  • brew install ffmpeg
  • brew install portaudio
  • pip install pyaudio
  • pip install pydub

Ensure a properly running SIP server has been configured on your network. Clients can hook into any existing SIP architecture as long as they are provided a conference number to dial into.

All testing was done with FreePBX/Asterisk 13. Setup instructions can be found here: https://wiki.freepbx.org/display/FOP/Installing+FreePBX+with+the+Official+Distro

Create SIP extensions (or allow anonymous inbound calls) for each user. These should have unique call numbers and passwords. The producer can then take these credentials and generate connection strings for each user. Role is determined and synchronized by the chat client and servers, so any SIP credentails will work for any type of user. They can be reused, but only one client can connect to the voice channels from one set of credentials at any time.

Some caveats: Installation behind NAT can be a bit trickey. If you only properly forward SIP ports you will be able to place phone calls but not actually connect to the voice channel. In order for this to occur, about 10,000 RTP ports will also need to be forwarded. Specific values will be noted in your server configuration. While there are some technologies to avoid having to forward all of these ports, they were not explored and may not be compatible without modifications to the clients (ICE/STUN).

If a client improperly exits, they will not be able to rejoin the session until a manual hangup is called on the server side or timeout has been reached. These timeouts are configurable on the server side and default to 30 munites under Asterisk 13.

After cloning the application, any desired configurations should be filled out in hqc/conn.conf:

• [ConnectionDetails]
  • user: username for the voice chat server
  • password: password for the voice chat server
  • server: voice chat server IP address
  • call_no: voice conference call number
• [AudioSettings]
  • mic: preferred audio recording device
  • speakers: preferred playback device
  • recording_location: This field should be a directory path where audio files from the recording sessions are saved.
• [ChatSettings]
  • ip_address: IP address of chat server
  • port: port address of chat server
  • username: desired username for the chat
  • role: role of user during session (set upon joining a session)
• [HQRecordingSettings]
  • width: TODO
  • channels: number of audio channels for high quality recording
  • rate: bit rate for high quality recording
• [LQRecordingSettings]
  • codec - desired audio compression

Clone the Application

• Make sure git is properly installed and included in your path
• git clone https://github.com/b6938236/high-quality-chat.git
• Install dependencies from section above

Running the Chat Server

• Follow instructions up until the server section on the chat README

Starting the Graphical Client

• Naviagte to project folder cd high-quality-chat
• And then run hqc/gui.py with python gui.py