9 from typing import Optional, List
14 gi.require_version('Gst', '1.0')
15 from gi.repository import Gst
17 gi.require_version('GstWebRTC', '1.0')
18 from gi.repository import GstWebRTC
20 gi.require_version('GstSdp', '1.0')
21 from gi.repository import GstSdp
23 gi.require_version('GstRtspServer', '1.0')
24 from gi.repository import Gst, GstRtspServer, GObject, GLib
26 log = logging.getLogger(__name__)
29 class GstreamerRtspServer():
31 server = GstRtspServer.RTSPServer()
32 server.set_address("::")
33 server.set_service('8554') # port as string
34 factory = GstRtspServer.RTSPMediaFactory()
35 factory.set_launch("intervideosrc ! decodebin ! theoraenc ! queue ! rtptheorapay name=pay0")
36 factory.set_shared(True)
37 mountPoints = server.get_mount_points()
38 mountPoints.add_factory("/cug", factory)
45 self.sdp_offer: Optional[str] = None
46 self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
47 self.session_id = None
48 self.received_ice_candidates = queue.Queue()
49 self.generated_ice_candidates = queue.Queue()
50 self.user_fragments: Optional[List] = None
51 self.mids: Optional[List] = None
55 def on_negotiation_needed(self, element):
56 log.debug('on_negotiation_needed')
58 def on_ice_candidate(self, element, mlineindex, candidate):
59 log.debug('on_ice_candidate')
60 self.generated_ice_candidates.put_nowait((mlineindex, candidate))
62 def webrtcbin_pad_added(self, element, pad):
63 log.debug('webrtcbin_pad_added')
64 if pad.direction != Gst.PadDirection.SRC:
66 decodebin = Gst.ElementFactory.make('decodebin')
67 decodebin.connect('pad-added', self.decodebin_pad_added)
68 self.pipe.add(decodebin)
69 decodebin.sync_state_with_parent()
70 self.webrtcbin.link(decodebin)
72 def decodebin_pad_added(self, element, pad):
73 log.debug('decodebin_pad_added')
74 if not pad.has_current_caps():
75 log.debug(pad, 'has no caps, ignoring')
78 caps = pad.get_current_caps()
79 padsize = caps.get_size()
80 for i in range(padsize):
81 s = caps.get_structure(i) # Gst.Structure
83 if name.startswith('video'):
84 q = Gst.ElementFactory.make('queue')
85 conv = Gst.ElementFactory.make('videoconvert')
86 sink = Gst.ElementFactory.make('intervideosink')
90 self.pipe.sync_children_states()
91 pad.link(q.get_static_pad('sink'))
94 elif name.startswith('audio'):
95 q = Gst.ElementFactory.make('queue')
96 conv = Gst.ElementFactory.make('audioconvert')
97 resample = Gst.ElementFactory.make('audioresample')
98 sink = Gst.ElementFactory.make('autoaudiosink')
101 self.pipe.add(resample)
103 self.pipe.sync_children_states()
104 pad.link(q.get_static_pad('sink'))
109 async def listen_to_gstreamer_bus(self):
110 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
111 self.pipe = Gst.Pipeline.new("pipeline")
112 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
113 bus = Gst.Pipeline.get_bus(self.pipe)
114 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
115 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
116 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
117 self.pipe.set_state(Gst.State.PLAYING)
120 if bus.have_pending():
122 if msg.type == Gst.MessageType.ERROR:
123 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
125 elif msg.type == Gst.MessageType.EOS: # end of stream
126 log.info(f'Gstreamer message bus reports end of stream')
128 elif self.sdp_offer is not None:
129 res, sm = GstSdp.SDPMessage.new()
130 assert res == GstSdp.SDPResult.OK
131 GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
132 # the three lines above can also be done this way in new versions of GStreamer:
133 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
134 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
135 gst_promise = Gst.Promise.new()
136 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
137 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
138 self.sdp_offer = None
140 gst_promise = Gst.Promise.new()
141 self.webrtcbin.emit('create-answer', None, gst_promise)
142 result = await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
143 assert result == Gst.PromiseResult.REPLIED
144 reply = gst_promise.get_reply()
145 answer = reply.get_value('answer')
146 sdp_message = answer.sdp
147 self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
148 for i in range(sdp_message.medias_len())]
149 self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
150 for i in range(sdp_message.medias_len())]
151 sdp_answer = sdp_message.as_text()
152 log.info(f'Send SDP answer:\n{sdp_answer}')
153 sdp_answer_msg = json.dumps({
154 'SessionID': self.session_id,
156 'Value': json.dumps({
161 gst_promise = Gst.Promise.new()
162 self.webrtcbin.emit('set-local-description', answer, gst_promise)
163 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
164 gst_promise.get_reply()
165 await self.websocket.send(sdp_answer_msg)
167 elif self.received_ice_candidates.qsize() > 0:
168 ic = self.received_ice_candidates.get_nowait()
169 if ic['candidate'] != '':
170 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
172 elif self.generated_ice_candidates.qsize() > 0:
173 mlineindex, candidate = self.generated_ice_candidates.get_nowait()
174 icemsg_value = json.dumps({
175 "candidate": candidate,
176 "sdpMid": self.mids[mlineindex],
177 "sdpMLineIndex": mlineindex,
178 "usernameFragment": self.user_fragments[mlineindex],
180 icemsg = json.dumps({
181 'SessionID': self.session_id,
182 'Type': 'addCalleeIceCandidate',
183 'Value': icemsg_value,
185 log.info(f'Send ICE candidate: {icemsg_value}')
186 await self.websocket.send(icemsg)
189 await asyncio.sleep(0.1)
191 self.pipe.set_state(Gst.State.NULL)
193 async def talk_to_websocket(self, uri):
194 async for msg in self.websocket:
195 msg_json = json.loads(msg)
196 msg_type = msg_json['Type']
197 msg_value = msg_json['Value']
198 assert self.session_id is None or self.session_id == msg_json['SessionID']
199 if msg_type == 'newSession':
200 self.session_id = msg_json['SessionID']
201 log.info(f"New session {self.session_id}")
202 elif msg_type == 'gotOffer':
203 value_json = json.loads(msg_value)
204 sdp = value_json['sdp']
205 log.info(f'Got SDP offer:\n{sdp}')
207 elif msg_type == 'addCallerIceCandidate':
208 value_json = json.loads(msg_value)
209 log.info(f'Got ICE candidate: {value_json}')
210 self.received_ice_candidates.put_nowait(value_json)
211 elif msg_type == 'roomNotFound':
212 log.error(f'The room was not found: {uri}')
214 elif msg_type == 'roomClosed':
215 log.info(f'Oh noes, the room went away (session {self.session_id})!')
216 self.session_id = None
219 log.error(f'Unknown message type {msg_type}')
221 async def run(self, uri):
222 ssl_context = ssl.SSLContext()
223 ssl_context.check_hostname = False
224 ssl_context.verify_mode = ssl.CERT_NONE
226 async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
227 talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
228 listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
229 main_loop = asyncio.Task(gstreamer_main_loop())
230 done, pending = await asyncio.wait(
231 [talk_to_websocket_task, listen_to_gstreamer_bus_task, main_loop],
232 return_when=asyncio.FIRST_COMPLETED)
241 async def gstreamer_main_loop():
242 """Does the equivalent of the following lines in an async friendly way:
243 loop = GLib.MainLoop()
246 gst_loop = GLib.MainLoop()
247 context = gst_loop.get_context()
249 events_dispatched = context.iteration(False)
250 await asyncio.sleep(0. if events_dispatched else 0.01)
254 logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
255 parser = argparse.ArgumentParser()
256 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
257 help='Signalling server URI')
258 args = parser.parse_args()
261 rtsp = GstreamerRtspServer()
263 asyncio.run(lagarde.run(args.uri), debug=True)
266 if __name__ == '__main__':