14 gi.require_version('Gst', '1.0')
15 from gi.repository import Gst
16 gi.require_version('GstWebRTC', '1.0')
17 from gi.repository import GstWebRTC
18 gi.require_version('GstSdp', '1.0')
19 from gi.repository import GstSdp
22 log = logging.getLogger(__name__)
27 def __init__(self, uri: str):
29 self.ssl_context = ssl.SSLContext()
30 self.ssl_context.check_hostname = False
31 self.ssl_context.verify_mode = ssl.CERT_NONE
33 self.session_id = None
35 def send_sdp_offer(self, offer):
36 text = offer.sdp.as_text()
37 log.info(f'send_sdp_offer with {text}')
39 'SessionID': self.session_id,
46 loop = asyncio.new_event_loop()
47 loop.run_until_complete(self.websocket.send(msg))
50 def on_offer_created(self, promise, _, __):
51 log.info('on_offer_created')
53 reply = promise.get_reply()
54 offer = reply['offer']
55 promise = Gst.Promise.new()
56 self.webrtc.emit('set-local-description', offer, promise)
58 self.send_sdp_offer(offer)
60 def on_negotiation_needed(self, element):
61 log.info('on_negotiation_needed')
62 promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
63 element.emit('create-offer', None, promise)
65 def send_ice_candidate_message(self, _, mlineindex, candidate):
67 'SessionID': self.session_id,
68 'Type': 'addCalleeIceCandidate',
70 "candidate": candidate,
72 "sdpMLineIndex": mlineindex,
75 log.info(f'send_ice_candidate_message with {icemsg}')
76 loop = asyncio.new_event_loop()
77 loop.run_until_complete(self.websocket.send(icemsg))
80 def on_incoming_decodebin_stream(self, _, pad):
81 log.info('on_incoming_decodebin_stream')
82 if not pad.has_current_caps():
83 log.info(pad, 'has no caps, ignoring')
86 caps = pad.get_current_caps()
90 if name.startswith('video'):
91 q = Gst.ElementFactory.make('queue')
92 conv = Gst.ElementFactory.make('videoconvert')
93 sink = Gst.ElementFactory.make('autovideosink')
94 self.pipe.add(q, conv, sink)
95 self.pipe.sync_children_states()
96 pad.link(q.get_static_pad('sink'))
99 elif name.startswith('audio'):
100 q = Gst.ElementFactory.make('queue')
101 conv = Gst.ElementFactory.make('audioconvert')
102 resample = Gst.ElementFactory.make('audioresample')
103 sink = Gst.ElementFactory.make('autoaudiosink')
104 self.pipe.add(q, conv, resample, sink)
105 self.pipe.sync_children_states()
106 pad.link(q.get_static_pad('sink'))
111 def on_incoming_stream(self, _, pad):
112 log.info('on_incoming_stream')
113 if pad.direction != Gst.PadDirection.SRC:
115 decodebin = Gst.ElementFactory.make('decodebin')
116 decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
117 self.pipe.add(decodebin)
118 decodebin.sync_state_with_parent()
119 self.webrtc.link(decodebin)
121 def start_pipeline(self):
122 self.webrtc = Gst.ElementFactory.make('webrtcbin', 'laplace')
123 self.webrtc.set_property("bundle-policy", 3)
124 direction = GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
125 caps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
126 self.webrtc.emit('add-transceiver', direction, caps)
127 self.pipe = Gst.Pipeline.new("pipeline")
128 Gst.Bin.do_add_element(self.pipe, self.webrtc)
129 self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
130 self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
131 self.webrtc.connect('pad-added', self.on_incoming_stream)
132 self.pipe.set_state(Gst.State.PLAYING)
134 def close_pipeline(self):
135 self.pipe.set_state(Gst.State.NULL)
139 def handle_sdp(self, sdp):
140 log.info(f'handle_sdp: {sdp}')
141 res, sdpmsg = GstSdp.SDPMessage.new()
142 GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
143 answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
144 promise = Gst.Promise.new()
145 self.webrtc.emit('set-remote-description', answer, promise)
148 def handle_ice(self, ice):
149 log.info(f'handle_ice: {ice}')
150 candidate = ice['candidate']
151 sdpmlineindex = ice['sdpMLineIndex']
152 self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
155 async with websockets.connect(self.uri, ssl=self.ssl_context) as websocket:
156 self.websocket = websocket
157 self.start_pipeline()
158 async for msg in websocket:
159 msg_json = json.loads(msg)
160 msg_type = msg_json['Type']
161 msg_value = msg_json['Value']
162 session_id = msg_json['SessionID']
163 log.info(f"receive for session {session_id} type {msg_type}")
164 if msg_type == 'newSession':
165 self.session_id = session_id
166 elif msg_type == 'gotOffer':
167 value_json = json.loads(msg_value)
168 sdp = value_json['sdp']
170 elif msg_type == 'addCallerIceCandidate':
171 value_json = json.loads(msg_value)
172 self.handle_ice(value_json)
173 self.close_pipeline()
174 self.websocket = None
175 self.session_id = None
179 for plugin in ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
180 "rtpmanager", "videotestsrc", "audiotestsrc"]:
181 if Gst.Registry.get().find_plugin(plugin) is None:
182 print('Missing gstreamer plugin:', plugin)
188 logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
190 if not check_plugins():
192 parser = argparse.ArgumentParser()
193 parser.add_argument('--uri', default='wss://localhost:2222/ws_connect?id=cug',
194 help='Signalling server URI')
195 args = parser.parse_args()
196 c = WebRTCClient(args.uri)
197 loop = asyncio.get_event_loop()
198 loop.run_until_complete(c.run())
201 if __name__=='__main__':