class WebRTCClient:
- def __init__(self, events: Events):
+ def __init__(self, events: Events, rtmp_uri: str):
self.events = events
+ self.rtmp_uri = rtmp_uri
self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
self.pipe = Gst.Pipeline.new("pipeline")
Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
flmux = Gst.ElementFactory.make('flvmux')
sink = Gst.ElementFactory.make('rtmpsink')
- sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa')
+ sink.set_property('location', self.rtmp_uri)
assert q and conv and enc and capsfilter and flmux and sink
self.pipe.add(q)
await asyncio.sleep(0.1)
-async def run(uri):
+async def run(laplace_uri: str, rtmp_uri: str):
try:
events = Events()
- webrtc = WebRTCClient(events)
- signaling = SignalingClient(events, uri)
+ webrtc = WebRTCClient(events, rtmp_uri)
+ signaling = SignalingClient(events, laplace_uri)
webrtc_task = asyncio.Task(webrtc.run())
signaling_task = asyncio.Task(signaling.run())
def main():
logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
+ default_source = 'wss://localhost:1234/ws_connect?id=cug'
+ default_dest = 'rtmp://localhost:1935/cug'
parser = argparse.ArgumentParser()
- parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
- help='Signalling server URI')
+ parser.add_argument('-s', '--source', default=default_source,
+ help=f'Laplace signalling websocket URI, default: {default_source}')
+ parser.add_argument('-d', '--destination', default=default_dest,
+ help=f'RTMP server URI, default: {default_dest}')
args = parser.parse_args()
Gst.init(None)
- asyncio.run(run(args.uri), debug=True)
+ asyncio.run(run(args.source, args.destination), debug=True)
if __name__ == '__main__':