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, rtmp_uri: str):
128 self.rtmp_uri = rtmp_uri
129 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
130 self.pipe = Gst.Pipeline.new("pipeline")
131 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
132 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
133 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
134 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
136 def on_negotiation_needed(self, element):
137 log.info('on_negotiation_needed')
139 def on_ice_candidate(self, element, mlineindex, candidate):
140 log.info('on_ice_candidate')
141 self.events.generated_ice_candidates.put_nowait((mlineindex, candidate))
143 def webrtcbin_pad_added(self, element, pad):
144 log.info('webrtcbin_pad_added')
145 if pad.direction != Gst.PadDirection.SRC:
147 decodebin = Gst.ElementFactory.make('decodebin')
148 decodebin.connect('pad-added', self.decodebin_pad_added)
149 self.pipe.add(decodebin)
150 decodebin.sync_state_with_parent()
151 self.webrtcbin.link(decodebin)
153 def decodebin_pad_added(self, element, pad):
154 log.info('decodebin_pad_added')
155 if not pad.has_current_caps():
156 log.info(pad, 'has no caps, ignoring')
158 caps = pad.get_current_caps()
159 padsize = caps.get_size()
161 for i in range(padsize):
162 s = caps.get_structure(i) # Gst.Structure
164 if name.startswith('video'):
165 q = Gst.ElementFactory.make('queue')
166 conv = Gst.ElementFactory.make('videoconvert')
167 enc = Gst.ElementFactory.make('x264enc')
168 enc.set_property('bitrate', 1000)
169 enc.set_property('tune', 'zerolatency')
170 capsfilter = Gst.ElementFactory.make('capsfilter')
171 capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
172 flmux = Gst.ElementFactory.make('flvmux')
173 sink = Gst.ElementFactory.make('rtmpsink')
174 sink.set_property('location', self.rtmp_uri)
175 assert q and conv and enc and capsfilter and flmux and sink
180 self.pipe.add(capsfilter)
184 q_pad_sink = q.get_static_pad('sink')
186 pad_link_return = pad.link(q_pad_sink)
187 assert pad_link_return == Gst.PadLinkReturn.OK
193 ok = enc.link(capsfilter)
195 ok = capsfilter.link(flmux)
197 ok = flmux.link(sink)
199 self.pipe.set_state(Gst.State.PLAYING)
200 self.pipe.sync_children_states()
202 elif name.startswith('audio'):
203 q = Gst.ElementFactory.make('queue')
204 conv = Gst.ElementFactory.make('audioconvert')
205 resample = Gst.ElementFactory.make('audioresample')
206 sink = Gst.ElementFactory.make('autoaudiosink')
209 self.pipe.add(resample)
211 self.pipe.sync_children_states()
212 pad.link(q.get_static_pad('sink'))
217 def set_remote_desciption_done(self, gst_promise):
218 gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done)
219 self.webrtcbin.emit('create-answer', None, gst_promise)
221 def create_answer_done(self, gst_promise):
222 reply = gst_promise.get_reply()
223 answer = reply.get_value('answer')
224 sdp_message = answer.sdp
225 mids = [sdp_message.get_media(i).get_attribute_val('mid')
226 for i in range(sdp_message.medias_len())]
227 user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
228 for i in range(sdp_message.medias_len())]
229 self.events.sdp_info.put_nowait((mids, user_fragments))
230 sdp_answer = sdp_message.as_text()
231 log.info(f'Send SDP answer')
232 log.debug(f'SDP answer:\n{sdp_answer}')
233 self.events.sdp_answer.put_nowait(sdp_answer)
234 gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
235 self.webrtcbin.emit('set-local-description', answer, gst_promise)
237 def set_local_description_done(self, gst_promise):
238 gst_promise.get_reply()
241 bus = Gst.Pipeline.get_bus(self.pipe)
242 self.pipe.set_state(Gst.State.PLAYING)
245 if bus.have_pending():
247 if msg.type == Gst.MessageType.ERROR:
248 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
250 elif msg.type == Gst.MessageType.EOS: # end of stream
251 log.info(f'Gstreamer message bus reports end of stream')
253 elif self.events.sdp_offer.qsize() > 0:
254 sdp_offer = self.events.sdp_offer.get_nowait()
255 res, sm = GstSdp.SDPMessage.new()
256 assert res == GstSdp.SDPResult.OK
257 GstSdp.sdp_message_parse_buffer(bytes(sdp_offer.encode()), sm)
258 # the three lines above can also be done this way in new versions of GStreamer:
259 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
260 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
261 gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done)
262 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
264 elif self.events.received_ice_candidates.qsize() > 0:
265 ic = self.events.received_ice_candidates.get_nowait()
266 if ic['candidate'] != '':
267 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
269 elif self.events.room_left.qsize() > 0:
270 self.events.room_left.get_nowait()
274 await asyncio.sleep(0.1)
276 self.pipe.set_state(Gst.State.NULL)
279 async def run_repeated(task):
282 await asyncio.sleep(0.1)
285 async def run(laplace_uri: str, rtmp_uri: str):
288 webrtc = WebRTCClient(events, rtmp_uri)
289 signaling = SignalingClient(events, laplace_uri)
291 webrtc_task = asyncio.Task(webrtc.run())
292 signaling_task = asyncio.Task(signaling.run())
294 done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED)
305 logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
306 default_source = 'wss://localhost:1234/ws_connect?id=cug'
307 default_dest = 'rtmp://localhost:1935/cug'
308 parser = argparse.ArgumentParser()
309 parser.add_argument('-s', '--source', default=default_source,
310 help=f'Laplace signalling websocket URI, default: {default_source}')
311 parser.add_argument('-d', '--destination', default=default_dest,
312 help=f'RTMP server URI, default: {default_dest}')
313 args = parser.parse_args()
316 asyncio.run(run(args.source, args.destination))
319 if __name__ == '__main__':