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("videotestsrc ! 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('autovideosink') # needs XDG_RUNTIME_DIR
87 sink = Gst.ElementFactory.make('xvimagesink')
91 self.pipe.sync_children_states()
92 pad.link(q.get_static_pad('sink'))
95 elif name.startswith('audio'):
96 q = Gst.ElementFactory.make('queue')
97 conv = Gst.ElementFactory.make('audioconvert')
98 resample = Gst.ElementFactory.make('audioresample')
99 sink = Gst.ElementFactory.make('autoaudiosink')
102 self.pipe.add(resample)
104 self.pipe.sync_children_states()
105 pad.link(q.get_static_pad('sink'))
110 async def listen_to_gstreamer_bus(self):
111 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
112 self.pipe = Gst.Pipeline.new("pipeline")
113 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
114 bus = Gst.Pipeline.get_bus(self.pipe)
115 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
116 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
117 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
118 self.pipe.set_state(Gst.State.PLAYING)
121 if bus.have_pending():
123 if msg.type == Gst.MessageType.ERROR:
124 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
126 elif msg.type == Gst.MessageType.EOS: # end of stream
127 log.info(f'Gstreamer message bus reports end of stream')
129 elif self.sdp_offer is not None:
130 res, sm = GstSdp.SDPMessage.new()
131 assert res == GstSdp.SDPResult.OK
132 GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
133 # the three lines above can also be done this way in new versions of GStreamer:
134 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
135 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
136 gst_promise = Gst.Promise.new()
137 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
138 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
139 self.sdp_offer = None
141 gst_promise = Gst.Promise.new()
142 self.webrtcbin.emit('create-answer', None, gst_promise)
143 result = await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
144 assert result == Gst.PromiseResult.REPLIED
145 reply = gst_promise.get_reply()
146 answer = reply.get_value('answer')
147 sdp_message = answer.sdp
148 self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
149 for i in range(sdp_message.medias_len())]
150 self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
151 for i in range(sdp_message.medias_len())]
152 sdp_answer = sdp_message.as_text()
153 log.info(f'Send SDP answer:\n{sdp_answer}')
154 sdp_answer_msg = json.dumps({
155 'SessionID': self.session_id,
157 'Value': json.dumps({
162 gst_promise = Gst.Promise.new()
163 self.webrtcbin.emit('set-local-description', answer, gst_promise)
164 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
165 gst_promise.get_reply()
166 await self.websocket.send(sdp_answer_msg)
168 elif self.received_ice_candidates.qsize() > 0:
169 ic = self.received_ice_candidates.get_nowait()
170 if ic['candidate'] != '':
171 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
173 elif self.generated_ice_candidates.qsize() > 0:
174 mlineindex, candidate = self.generated_ice_candidates.get_nowait()
175 icemsg_value = json.dumps({
176 "candidate": candidate,
177 "sdpMid": self.mids[mlineindex],
178 "sdpMLineIndex": mlineindex,
179 "usernameFragment": self.user_fragments[mlineindex],
181 icemsg = json.dumps({
182 'SessionID': self.session_id,
183 'Type': 'addCalleeIceCandidate',
184 'Value': icemsg_value,
186 log.info(f'Send ICE candidate: {icemsg_value}')
187 await self.websocket.send(icemsg)
190 await asyncio.sleep(0.1)
192 self.pipe.set_state(Gst.State.NULL)
194 async def talk_to_websocket(self, uri):
195 async for msg in self.websocket:
196 msg_json = json.loads(msg)
197 msg_type = msg_json['Type']
198 msg_value = msg_json['Value']
199 assert self.session_id is None or self.session_id == msg_json['SessionID']
200 if msg_type == 'newSession':
201 self.session_id = msg_json['SessionID']
202 log.info(f"New session {self.session_id}")
203 elif msg_type == 'gotOffer':
204 value_json = json.loads(msg_value)
205 sdp = value_json['sdp']
206 log.info(f'Got SDP offer:\n{sdp}')
208 elif msg_type == 'addCallerIceCandidate':
209 value_json = json.loads(msg_value)
210 log.info(f'Got ICE candidate: {value_json}')
211 self.received_ice_candidates.put_nowait(value_json)
212 elif msg_type == 'roomNotFound':
213 log.error(f'The room was not found: {uri}')
215 elif msg_type == 'roomClosed':
216 log.info(f'Oh noes, the room went away (session {self.session_id})!')
217 self.session_id = None
220 log.error(f'Unknown message type {msg_type}')
222 async def run(self, uri):
223 ssl_context = ssl.SSLContext()
224 ssl_context.check_hostname = False
225 ssl_context.verify_mode = ssl.CERT_NONE
227 async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
228 talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
229 listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
230 main_loop = asyncio.Task(gstreamer_main_loop())
231 done, pending = await asyncio.wait(
232 [talk_to_websocket_task, listen_to_gstreamer_bus_task, main_loop],
233 return_when=asyncio.FIRST_COMPLETED)
242 async def gstreamer_main_loop():
243 """Does the equivalent of the following lines in an async friendly way:
244 loop = GLib.MainLoop()
247 gst_loop = GLib.MainLoop()
248 context = gst_loop.get_context()
250 events_dispatched = context.iteration(False)
251 await asyncio.sleep(0. if events_dispatched else 0.01)
255 logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
256 parser = argparse.ArgumentParser()
257 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
258 help='Signalling server URI')
259 args = parser.parse_args()
262 rtsp = GstreamerRtspServer()
264 asyncio.run(lagarde.run(args.uri), debug=True)
267 if __name__ == '__main__':