janus-client 0.2.5

Janus webrtc client in async Python.

Janus Client in Python

This is a Janus webrtc client written in Python, to be used with asyncio.

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Install

pip install janus-client

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Description

The package is implementing a general purpose client that can communicate with a Janus server.

Examples like VideoRoom plugin are also included in the package, but currently it depends on GStreamer for WebRTC and media streaming, and it will not be automatically installed. The reason for this is because it's not trivial to install/recompile it. Please refer to Quirks section.

The VideoRoom plugin implemented in plugin_video_room_ffmpeg.py uses FFmpeg instead. It depends on the ffmpeg-cli, but that is required to be installed separately.

FFmpeg Stream To WebRTC (WARNING !!!)

This FFmpeg stream to WebRTC solution is a hacked solution. The fact is that FFmpeg doesn't support WebRTC and aiortc is implemented using PyAV. PyAV has much less features than a full fledged installed FFmpeg, so to support more features and keep things simple, I hacked about a solution without the use of WHIP server or UDP of RTMP.

First the ffmpeg input part should be constructed by the user, before passing it to janus_client.media.MediaPlayer. When the media player needs to stream the video, the following happens:

• a thread will be created
• a ffmpeg process will be created. Output of ffmpeg is hardcode to be rawvideo rgb24
• thread reads output of ffmpeg process
• coverts the output data to numpy array
• use PyAV to convert the numpy array to av.VideoFrame frame
• hack the pts and time_base parameter of the frame. I don't know what it is and just found a value that works.
• put the frame into video track queue as that is what is required by aiortc.mediastreams.MediaStreamTrack.

Next the MediaPlayer is used as required by aiortc. An example of that can be found here.

A fork of FFmpeg that supports WebRTC can be found here.

Features

:heavy_check_mark: Connect to Janus server through websocket (using websockets)
:heavy_check_mark: Handle transactions with Janus
:heavy_check_mark: Create/destroy sessions
:heavy_check_mark: Create/destroy plugins
:heavy_check_mark: Handle multiple sessions and or multiple plugins at the same time
:heavy_check_mark: Support authentication with shared static secret (API key) and/or stored token
:heavy_check_mark: Expose Admin/Monitor API client

In Progress

:clock3: Emit events to respective session and plugin handlers
:clock3: Create plugin for videoroom plugin using GStreamer
:clock3: Documentation

Dependencies

• websockets
• aiortc
• ffmpeg-python
• numpy
• av

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Usage

In test_gst_videoroom.py, you will see an example on how to use the client in general, such as connecting and creating sessions.
I use it to develop this package.

For more details, please Read The Docs

Example:

import asyncio
import ssl
import pathlib
from janus_client import JanusClient
from janus_client.plugin_video_room import JanusVideoRoomPlugin

ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_CLIENT)
localhost_pem = pathlib.Path(__file__).with_name("lt_limmengkiat_name_my.crt")
ssl_context.load_verify_locations(localhost_pem)

async def main():
    # Connect to server
    client = JanusClient("wss://lt.limmengkiat.name.my/janusws/")
    await client.connect(ssl=ssl_context)
    # Create session
    session = await client.create_session()
    # Create plugin
    plugin_handle = await session.create_plugin_handle(JanusVideoRoomPlugin)

    participants = await plugin_handle.list_participants(1234)
    if len(participants) > 0:
        # Publishers available
        participants_data_1 = participants[0]
        participant_id = participants_data_1["id"]

        # Subscribe to publisher, will get jsep (sdp offer)
        await plugin_handle.subscribe(1234, participant_id)
        # WebRTC streaming not implemented  for subscriber yet
        await asyncio.sleep(5)
        # Unsubscribe from the publisher
        await plugin_handle.unsubscribe()

    # Destroy plugin
    await plugin_handle.destroy()
    # Destroy session
    await session.destroy()
    # Destroy connection
    await client.disconnect()

asyncio.run(main())

Quirks

1. On my RPI 2 Raspbian Buster, there's a problem with GStreamer installed from distribution repository. It's complaining about ssl and then failing DTLS.
   Referring to this PR: webrtcbin: fix DTLS when receivebin is set to DROP
   I believe there is a bug in the distributed GStreamer version (v1.14.4) thus I recompiled it on my RPI 2
   There's also a chance that it's a problem with openssl itself, an incompatibility. Refering to this gist: OpenSSL DTLS problem in Debian buster

   Because of these, please recompile GStreamer with version above 1.14.4.

2. When recompiling GStreamer on RPI, there's this issue: rpicamsrc 1.18.3 failed.
   You can patch the build with this PR: rpicamsrc: depend on posix threads and vchiq_arm or build with master branch.

3. The example was still unable to setup a peer connection to my janus server at lt.limmengkiat.name.my. I had to enable ice_tcp (ice_tcp=true) in janus.jcfg for it to work. I don't know why yet.
   

Recompiling GStreamer

Please refer to our Wiki page: Compiling GStreamer