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_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! theoraenc ! queue ! rtptheorapay name=pay0")
37 # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,format=I420,framerate=10/1 ! theoraenc ! queue ! rtptheorapay name=pay0")
38 # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
39 factory.set_launch("intervideosrc ! decodebin ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
40 # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,framerate=10/1 ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
41 factory.set_shared(True)
42 mountPoints = server.get_mount_points()
43 mountPoints.add_factory("/cug", factory)
50 self.sdp_offer: Optional[str] = None
51 self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
52 self.session_id = None
53 self.received_ice_candidates = queue.Queue()
54 self.generated_ice_candidates = queue.Queue()
55 self.user_fragments: Optional[List] = None
56 self.mids: Optional[List] = None
60 def on_negotiation_needed(self, element):
61 log.debug('on_negotiation_needed')
63 def on_ice_candidate(self, element, mlineindex, candidate):
64 log.debug('on_ice_candidate')
65 self.generated_ice_candidates.put_nowait((mlineindex, candidate))
67 def webrtcbin_pad_added(self, element, pad):
68 log.debug('webrtcbin_pad_added')
69 if pad.direction != Gst.PadDirection.SRC:
71 decodebin = Gst.ElementFactory.make('decodebin')
72 decodebin.connect('pad-added', self.decodebin_pad_added)
73 self.pipe.add(decodebin)
74 decodebin.sync_state_with_parent()
75 self.webrtcbin.link(decodebin)
77 def decodebin_pad_added(self, element, pad):
78 log.debug('decodebin_pad_added')
79 if not pad.has_current_caps():
80 log.debug(pad, 'has no caps, ignoring')
83 caps = pad.get_current_caps()
84 padsize = caps.get_size()
85 for i in range(padsize):
86 s = caps.get_structure(i) # Gst.Structure
88 if name.startswith('video'):
89 q = Gst.ElementFactory.make('queue')
90 conv = Gst.ElementFactory.make('videoconvert')
91 sink = Gst.ElementFactory.make('intervideosink')
95 self.pipe.sync_children_states()
96 pad.link(q.get_static_pad('sink'))
99 self.pipe.set_state(Gst.State.PLAYING)
101 async def listen_to_gstreamer_bus(self):
102 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
103 self.pipe = Gst.Pipeline.new("pipeline")
104 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
105 bus = Gst.Pipeline.get_bus(self.pipe)
106 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
107 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
108 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
109 self.pipe.set_state(Gst.State.PLAYING)
112 if bus.have_pending():
114 if msg.type == Gst.MessageType.ERROR:
115 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
117 elif msg.type == Gst.MessageType.EOS: # end of stream
118 log.info(f'Gstreamer message bus reports end of stream')
120 elif self.sdp_offer is not None:
121 res, sm = GstSdp.SDPMessage.new()
122 assert res == GstSdp.SDPResult.OK
123 GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
124 # the three lines above can also be done this way in new versions of GStreamer:
125 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
126 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
127 gst_promise = Gst.Promise.new()
128 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
129 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
130 self.sdp_offer = None
132 gst_promise = Gst.Promise.new()
133 self.webrtcbin.emit('create-answer', None, gst_promise)
134 result = await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
135 assert result == Gst.PromiseResult.REPLIED
136 reply = gst_promise.get_reply()
137 answer = reply.get_value('answer')
138 sdp_message = answer.sdp
139 self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
140 for i in range(sdp_message.medias_len())]
141 self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
142 for i in range(sdp_message.medias_len())]
143 sdp_answer = sdp_message.as_text()
144 log.info(f'Send SDP answer:\n{sdp_answer}')
145 sdp_answer_msg = json.dumps({
146 'SessionID': self.session_id,
148 'Value': json.dumps({
153 gst_promise = Gst.Promise.new()
154 self.webrtcbin.emit('set-local-description', answer, gst_promise)
155 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
156 gst_promise.get_reply()
157 await self.websocket.send(sdp_answer_msg)
159 elif self.received_ice_candidates.qsize() > 0:
160 ic = self.received_ice_candidates.get_nowait()
161 if ic['candidate'] != '':
162 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
164 elif self.generated_ice_candidates.qsize() > 0:
165 mlineindex, candidate = self.generated_ice_candidates.get_nowait()
166 icemsg_value = json.dumps({
167 "candidate": candidate,
168 "sdpMid": self.mids[mlineindex],
169 "sdpMLineIndex": mlineindex,
170 "usernameFragment": self.user_fragments[mlineindex],
172 icemsg = json.dumps({
173 'SessionID': self.session_id,
174 'Type': 'addCalleeIceCandidate',
175 'Value': icemsg_value,
177 log.info(f'Send ICE candidate: {icemsg_value}')
178 await self.websocket.send(icemsg)
181 await asyncio.sleep(0.1)
183 self.pipe.set_state(Gst.State.NULL)
185 async def talk_to_websocket(self, uri, ssl_context):
186 async with websockets.connect(uri, ssl=ssl_context, close_timeout=0.5) as self.websocket:
187 async for msg in self.websocket:
188 msg_json = json.loads(msg)
189 msg_type = msg_json['Type']
190 msg_value = msg_json['Value']
191 assert self.session_id is None or self.session_id == msg_json['SessionID']
192 if msg_type == 'newSession':
193 self.session_id = msg_json['SessionID']
194 log.info(f"New session {self.session_id}")
195 elif msg_type == 'gotOffer':
196 value_json = json.loads(msg_value)
197 sdp = value_json['sdp']
198 log.info(f'Got SDP offer:\n{sdp}')
200 elif msg_type == 'addCallerIceCandidate':
201 value_json = json.loads(msg_value)
202 log.info(f'Got ICE candidate: {value_json}')
203 self.received_ice_candidates.put_nowait(value_json)
204 elif msg_type == 'roomNotFound':
205 log.error(f'The room was not found: {uri}')
207 elif msg_type == 'roomClosed':
208 log.info(f'Oh noes, the room went away (session {self.session_id})!')
209 self.session_id = None
212 log.error(f'Unknown message type {msg_type}')
214 async def talk_to_signaling_server(self, uri):
215 ssl_context = ssl.SSLContext()
216 ssl_context.check_hostname = False
217 ssl_context.verify_mode = ssl.CERT_NONE
219 await self.talk_to_websocket(uri, ssl_context)
220 await asyncio.sleep(0.1)
222 async def run(self, uri):
224 talk_to_signaling_server_task = asyncio.Task(self.talk_to_signaling_server(uri))
225 listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
226 main_loop = asyncio.Task(gstreamer_main_loop())
227 done, pending = await asyncio.wait(
228 [talk_to_signaling_server_task, listen_to_gstreamer_bus_task, main_loop],
229 return_when=asyncio.FIRST_COMPLETED)
238 async def gstreamer_main_loop():
239 """Does the equivalent of the following lines in an async friendly way:
240 loop = GLib.MainLoop()
243 gst_loop = GLib.MainLoop()
244 context = gst_loop.get_context()
246 events_dispatched = context.iteration(False)
247 await asyncio.sleep(0. if events_dispatched else 0.01)
251 logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
252 parser = argparse.ArgumentParser()
253 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
254 help='Signalling server URI')
255 args = parser.parse_args()
258 rtsp = GstreamerRtspServer()
260 asyncio.run(lagarde.run(args.uri), debug=True)
263 if __name__ == '__main__':