12 gi.require_version('Gst', '1.0')
13 from gi.repository import Gst
15 gi.require_version('GstWebRTC', '1.0')
16 from gi.repository import GstWebRTC
18 gi.require_version('GstSdp', '1.0')
19 from gi.repository import GstSdp
21 log = logging.getLogger(__name__)
26 self.sdp_offer = queue.Queue()
27 self.sdp_answer = queue.Queue()
28 self.generated_ice_candidates = queue.Queue()
29 self.received_ice_candidates = queue.Queue()
30 self.sdp_info = queue.Queue() # (sdp_mids, user_fragments)
31 self.room_left = queue.Queue()
34 class SignalingClient:
35 def __init__(self, events: Events, uri):
38 self.ssl_context = ssl.SSLContext()
39 self.ssl_context.check_hostname = False
40 self.ssl_context.verify_mode = ssl.CERT_NONE
41 self.session_id = None
43 async def receive(self, uri):
44 async for msg in self.websocket:
45 msg_json = json.loads(msg)
46 msg_type = msg_json['Type']
47 msg_value = msg_json['Value']
48 assert self.session_id is None or self.session_id == msg_json['SessionID']
49 if msg_type == 'newSession':
50 self.session_id = msg_json['SessionID']
51 log.info(f"New session {self.session_id}")
52 elif msg_type == 'gotOffer':
53 value_json = json.loads(msg_value)
54 sdp = value_json['sdp']
55 log.info(f'Got SDP offer')
56 log.debug(f'SDP offer:\n{sdp}')
57 self.events.sdp_offer.put_nowait(sdp)
58 elif msg_type == 'addCallerIceCandidate':
59 value_json = json.loads(msg_value)
60 log.info(f'Got ICE candidate')
61 log.debug(f'ICE candidate: {value_json}')
62 self.events.received_ice_candidates.put_nowait(value_json)
63 elif msg_type == 'roomNotFound':
64 log.error(f'The room was not found: {uri}')
66 elif msg_type == 'roomClosed':
67 log.info(f'Oh noes, the room went away (session {self.session_id})!')
68 self.events.room_left.put_nowait(True)
71 log.error(f'Unknown message type {msg_type}')
77 if self.events.sdp_answer.qsize() > 0:
78 sdp_answer = self.events.sdp_answer.get_nowait()
79 sdp_answer_msg = json.dumps({
80 'SessionID': self.session_id,
87 await self.websocket.send(sdp_answer_msg)
89 elif self.events.sdp_info.qsize() > 0:
90 sdp_mids, user_fragments = self.events.sdp_info.get_nowait()
92 elif self.events.generated_ice_candidates.qsize() > 0 \
93 and sdp_mids is not None and user_fragments is not None:
94 mlineindex, candidate = self.events.generated_ice_candidates.get_nowait()
95 sdp_mid = sdp_mids[mlineindex]
96 user_fragment = user_fragments[mlineindex]
97 icemsg_value = json.dumps({
98 "candidate": candidate,
100 "sdpMLineIndex": mlineindex,
101 "usernameFragment": user_fragment,
103 icemsg = json.dumps({
104 'SessionID': self.session_id,
105 'Type': 'addCalleeIceCandidate',
106 'Value': icemsg_value,
108 log.info(f'Send ICE candidate')
109 log.debug(f'ICE candidate: {icemsg_value}')
110 await self.websocket.send(icemsg)
113 await asyncio.sleep(0.2)
116 self.session_id = None
117 async with websockets.connect(self.uri, ssl=self.ssl_context, close_timeout=0.5) as self.websocket:
118 receive_task = asyncio.Task(self.receive(self.uri))
119 send_task = asyncio.Task(self.send())
120 done, pending = await asyncio.wait([receive_task, send_task], return_when=asyncio.FIRST_COMPLETED)
126 def __init__(self, events: Events):
128 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
129 self.pipe = Gst.Pipeline.new("pipeline")
130 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
131 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
132 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
133 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
135 def on_negotiation_needed(self, element):
136 log.info('on_negotiation_needed')
138 def on_ice_candidate(self, element, mlineindex, candidate):
139 log.info('on_ice_candidate')
140 self.events.generated_ice_candidates.put_nowait((mlineindex, candidate))
142 def webrtcbin_pad_added(self, element, pad):
143 log.info('webrtcbin_pad_added')
144 if pad.direction != Gst.PadDirection.SRC:
146 decodebin = Gst.ElementFactory.make('decodebin')
147 decodebin.connect('pad-added', self.decodebin_pad_added)
148 self.pipe.add(decodebin)
149 decodebin.sync_state_with_parent()
150 self.webrtcbin.link(decodebin)
152 def decodebin_pad_added(self, element, pad):
153 log.info('decodebin_pad_added')
154 if not pad.has_current_caps():
155 log.info(pad, 'has no caps, ignoring')
157 caps = pad.get_current_caps()
158 padsize = caps.get_size()
160 for i in range(padsize):
161 s = caps.get_structure(i) # Gst.Structure
163 if name.startswith('video'):
164 q = Gst.ElementFactory.make('queue')
165 conv = Gst.ElementFactory.make('videoconvert')
166 enc = Gst.ElementFactory.make('x264enc')
167 enc.set_property('bitrate', 1000)
168 enc.set_property('tune', 'zerolatency')
169 capsfilter = Gst.ElementFactory.make('capsfilter')
170 capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
171 flmux = Gst.ElementFactory.make('flvmux')
172 sink = Gst.ElementFactory.make('rtmpsink')
173 sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa')
174 assert q and conv and enc and capsfilter and flmux and sink
179 self.pipe.add(capsfilter)
183 q_pad_sink = q.get_static_pad('sink')
185 pad_link_return = pad.link(q_pad_sink)
186 assert pad_link_return == Gst.PadLinkReturn.OK
192 ok = enc.link(capsfilter)
194 ok = capsfilter.link(flmux)
196 ok = flmux.link(sink)
198 self.pipe.set_state(Gst.State.PLAYING)
199 self.pipe.sync_children_states()
201 elif name.startswith('audio'):
202 q = Gst.ElementFactory.make('queue')
203 conv = Gst.ElementFactory.make('audioconvert')
204 resample = Gst.ElementFactory.make('audioresample')
205 sink = Gst.ElementFactory.make('autoaudiosink')
208 self.pipe.add(resample)
210 self.pipe.sync_children_states()
211 pad.link(q.get_static_pad('sink'))
216 def set_remote_desciption_done(self, gst_promise):
217 gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done)
218 self.webrtcbin.emit('create-answer', None, gst_promise)
220 def create_answer_done(self, gst_promise):
221 reply = gst_promise.get_reply()
222 answer = reply.get_value('answer')
223 sdp_message = answer.sdp
224 mids = [sdp_message.get_media(i).get_attribute_val('mid')
225 for i in range(sdp_message.medias_len())]
226 user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
227 for i in range(sdp_message.medias_len())]
228 self.events.sdp_info.put_nowait((mids, user_fragments))
229 sdp_answer = sdp_message.as_text()
230 log.info(f'Send SDP answer')
231 log.debug(f'SDP answer:\n{sdp_answer}')
232 self.events.sdp_answer.put_nowait(sdp_answer)
233 gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
234 self.webrtcbin.emit('set-local-description', answer, gst_promise)
236 def set_local_description_done(self, gst_promise):
237 gst_promise.get_reply()
240 bus = Gst.Pipeline.get_bus(self.pipe)
241 self.pipe.set_state(Gst.State.PLAYING)
244 if bus.have_pending():
246 if msg.type == Gst.MessageType.ERROR:
247 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
249 elif msg.type == Gst.MessageType.EOS: # end of stream
250 log.info(f'Gstreamer message bus reports end of stream')
252 elif self.events.sdp_offer.qsize() > 0:
253 sdp_offer = self.events.sdp_offer.get_nowait()
254 res, sm = GstSdp.SDPMessage.new()
255 assert res == GstSdp.SDPResult.OK
256 GstSdp.sdp_message_parse_buffer(bytes(sdp_offer.encode()), sm)
257 # the three lines above can also be done this way in new versions of GStreamer:
258 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
259 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
260 gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done)
261 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
263 elif self.events.received_ice_candidates.qsize() > 0:
264 ic = self.events.received_ice_candidates.get_nowait()
265 if ic['candidate'] != '':
266 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
268 elif self.events.room_left.qsize() > 0:
269 self.events.room_left.get_nowait()
273 await asyncio.sleep(0.1)
275 self.pipe.set_state(Gst.State.NULL)
278 async def run_repeated(task):
281 await asyncio.sleep(0.1)
287 webrtc = WebRTCClient(events)
288 signaling = SignalingClient(events, uri)
290 webrtc_task = asyncio.Task(webrtc.run())
291 signaling_task = asyncio.Task(signaling.run())
293 done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED)
304 logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
305 parser = argparse.ArgumentParser()
306 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
307 help='Signalling server URI')
308 args = parser.parse_args()
311 asyncio.run(run(args.uri), debug=True)
314 if __name__ == '__main__':