# Janus Videoroom example # Copyright @tobiasfriden and @saket424 on github # See https://github.com/centricular/gstwebrtc-demos/issues/66 # Copyright Jan Schmidt 2020 import random import ssl import websockets import asyncio import os import sys import json import argparse import string from websockets.exceptions import ConnectionClosed import attr # Set to False to send H.264 DO_VP8 = True # Set to False to disable RTX (lost packet retransmission) DO_RTX = True # Choose the video source: VIDEO_SRC="videotestsrc pattern=ball" # VIDEO_SRC="v4l2src" @attr.s class JanusEvent: sender = attr.ib(validator=attr.validators.instance_of(int)) @attr.s class PluginData(JanusEvent): plugin = attr.ib(validator=attr.validators.instance_of(str)) data = attr.ib() jsep = attr.ib() @attr.s class WebrtcUp(JanusEvent): pass @attr.s class Media(JanusEvent): receiving = attr.ib(validator=attr.validators.instance_of(bool)) kind = attr.ib(validator=attr.validators.in_(["audio", "video"])) @kind.validator def validate_kind(self, attribute, kind): if kind not in ["video", "audio"]: raise ValueError("kind must equal video or audio") @attr.s class SlowLink(JanusEvent): uplink = attr.ib(validator=attr.validators.instance_of(bool)) lost = attr.ib(validator=attr.validators.instance_of(int)) @attr.s class HangUp(JanusEvent): reason = attr.ib(validator=attr.validators.instance_of(str)) @attr.s(cmp=False) class Ack: transaction = attr.ib(validator=attr.validators.instance_of(str)) @attr.s class Jsep: sdp = attr.ib() type = attr.ib(validator=attr.validators.in_(["offer", "pranswer", "answer", "rollback"])) import gi gi.require_version('Gst', '1.0') from gi.repository import Gst gi.require_version('GstWebRTC', '1.0') from gi.repository import GstWebRTC gi.require_version('GstSdp', '1.0') from gi.repository import GstSdp if DO_VP8: ( encoder, payloader, rtp_encoding) = ( "vp8enc target-bitrate=100000 overshoot=25 undershoot=100 deadline=33000 keyframe-max-dist=1", "rtpvp8pay picture-id-mode=2", "VP8" ) else: ( encoder, payloader, rtp_encoding) = ( "x264enc", "rtph264pay", "H264" ) PIPELINE_DESC = ''' webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302 {} ! video/x-raw,width=640,height=480 ! videoconvert ! queue ! {} ! {} ! queue ! application/x-rtp,media=video,encoding-name={},payload=96 ! sendrecv. '''.format(VIDEO_SRC, encoder, payloader, rtp_encoding) def transaction_id(): return ''.join(random.choice(string.ascii_uppercase + string.digits) for _ in range(8)) @attr.s class JanusGateway: server = attr.ib(validator=attr.validators.instance_of(str)) #secure = attr.ib(default=True) _messages = attr.ib(factory=set) conn = None async def connect(self): sslCon=None if self.server.startswith("wss"): sslCon=ssl.SSLContext() self.conn = await websockets.connect(self.server, subprotocols=['janus-protocol'], ssl=sslCon) transaction = transaction_id() await self.conn.send(json.dumps({ "janus": "create", "transaction": transaction })) resp = await self.conn.recv() print (resp) parsed = json.loads(resp) assert parsed["janus"] == "success", "Failed creating session" assert parsed["transaction"] == transaction, "Incorrect transaction" self.session = parsed["data"]["id"] async def close(self): if self.conn: await self.conn.close() async def attach(self, plugin): assert hasattr(self, "session"), "Must connect before attaching to plugin" transaction = transaction_id() await self.conn.send(json.dumps({ "janus": "attach", "session_id": self.session, "plugin": plugin, "transaction": transaction })) resp = await self.conn.recv() parsed = json.loads(resp) assert parsed["janus"] == "success", "Failed attaching to {}".format(plugin) assert parsed["transaction"] == transaction, "Incorrect transaction" self.handle = parsed["data"]["id"] async def sendtrickle(self, candidate): assert hasattr(self, "session"), "Must connect before sending messages" assert hasattr(self, "handle"), "Must attach before sending messages" transaction = transaction_id() janus_message = { "janus": "trickle", "session_id": self.session, "handle_id": self.handle, "transaction": transaction, "candidate": candidate } await self.conn.send(json.dumps(janus_message)) #while True: # resp = await self._recv_and_parse() # if isinstance(resp, PluginData): # return resp # else: # self._messages.add(resp) # async def sendmessage(self, body, jsep=None): assert hasattr(self, "session"), "Must connect before sending messages" assert hasattr(self, "handle"), "Must attach before sending messages" transaction = transaction_id() janus_message = { "janus": "message", "session_id": self.session, "handle_id": self.handle, "transaction": transaction, "body": body } if jsep is not None: janus_message["jsep"] = jsep await self.conn.send(json.dumps(janus_message)) #while True: # resp = await self._recv_and_parse() # if isinstance(resp, PluginData): # if jsep is not None: # await client.handle_sdp(resp.jsep) # return resp # else: # self._messages.add(resp) async def keepalive(self): assert hasattr(self, "session"), "Must connect before sending messages" assert hasattr(self, "handle"), "Must attach before sending messages" while True: try: await asyncio.sleep(10) transaction = transaction_id() await self.conn.send(json.dumps({ "janus": "keepalive", "session_id": self.session, "handle_id": self.handle, "transaction": transaction })) except KeyboardInterrupt: return async def recv(self): if len(self._messages) > 0: return self._messages.pop() else: return await self._recv_and_parse() async def _recv_and_parse(self): raw = json.loads(await self.conn.recv()) janus = raw["janus"] if janus == "event": return PluginData( sender=raw["sender"], plugin=raw["plugindata"]["plugin"], data=raw["plugindata"]["data"], jsep=raw["jsep"] if "jsep" in raw else None ) elif janus == "webrtcup": return WebrtcUp( sender=raw["sender"] ) elif janus == "media": return Media( sender=raw["sender"], receiving=raw["receiving"], kind=raw["type"] ) elif janus == "slowlink": return SlowLink( sender=raw["sender"], uplink=raw["uplink"], lost=raw["lost"] ) elif janus == "hangup": return HangUp( sender=raw["sender"], reason=raw["reason"] ) elif janus == "ack": return Ack( transaction=raw["transaction"] ) else: return raw class WebRTCClient: def __init__(self, peer_id, server): self.conn = None self.pipe = None self.webrtc = None self.peer_id = peer_id self.signaling = JanusGateway(server) self.request = None self.offermsg = None def send_sdp_offer(self, offer): text = offer.sdp.as_text() print ('Sending offer:\n%s' % text) # configure media media = {'audio': True, 'video': True} request = {'request': 'publish'} request.update(media) self.request = request self.offermsg = { 'sdp': text, 'trickle': True, 'type': 'offer' } print (self.offermsg) loop = asyncio.new_event_loop() loop.run_until_complete(self.signaling.sendmessage(self.request, self.offermsg)) def on_offer_created(self, promise, _, __): promise.wait() reply = promise.get_reply() offer = reply.get_value('offer') promise = Gst.Promise.new() self.webrtc.emit('set-local-description', offer, promise) promise.interrupt() self.send_sdp_offer(offer) def on_negotiation_needed(self, element): promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None) element.emit('create-offer', None, promise) def send_ice_candidate_message(self, _, mlineindex, candidate): icemsg = {'candidate': candidate, 'sdpMLineIndex': mlineindex} print ("Sending ICE", icemsg) loop = asyncio.new_event_loop() loop.run_until_complete(self.signaling.sendtrickle(icemsg)) def on_incoming_decodebin_stream(self, _, pad): if not pad.has_current_caps(): print (pad, 'has no caps, ignoring') return caps = pad.get_current_caps() name = caps.to_string() if name.startswith('video'): q = Gst.ElementFactory.make('queue') conv = Gst.ElementFactory.make('videoconvert') sink = Gst.ElementFactory.make('autovideosink') self.pipe.add(q) self.pipe.add(conv) self.pipe.add(sink) self.pipe.sync_children_states() pad.link(q.get_static_pad('sink')) q.link(conv) conv.link(sink) elif name.startswith('audio'): q = Gst.ElementFactory.make('queue') conv = Gst.ElementFactory.make('audioconvert') resample = Gst.ElementFactory.make('audioresample') sink = Gst.ElementFactory.make('autoaudiosink') self.pipe.add(q) self.pipe.add(conv) self.pipe.add(resample) self.pipe.add(sink) self.pipe.sync_children_states() pad.link(q.get_static_pad('sink')) q.link(conv) conv.link(resample) resample.link(sink) def on_incoming_stream(self, _, pad): if pad.direction != Gst.PadDirection.SRC: return decodebin = Gst.ElementFactory.make('decodebin') decodebin.connect('pad-added', self.on_incoming_decodebin_stream) self.pipe.add(decodebin) decodebin.sync_state_with_parent() self.webrtc.link(decodebin) def start_pipeline(self): self.pipe = Gst.parse_launch(PIPELINE_DESC) self.webrtc = self.pipe.get_by_name('sendrecv') self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed) self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message) self.webrtc.connect('pad-added', self.on_incoming_stream) trans = self.webrtc.emit('get-transceiver', 0) if DO_RTX: trans.set_property ('do-nack', True) self.pipe.set_state(Gst.State.PLAYING) def extract_ice_from_sdp(self, sdp): mlineindex = -1 for line in sdp.splitlines(): if line.startswith("a=candidate"): candidate = line[2:] if mlineindex < 0: print("Received ice candidate in SDP before any m= line") continue print ('Received remote ice-candidate mlineindex {}: {}'.format(mlineindex, candidate)) self.webrtc.emit('add-ice-candidate', mlineindex, candidate) elif line.startswith("m="): mlineindex += 1 async def handle_sdp(self, msg): print (msg) if 'sdp' in msg: sdp = msg['sdp'] assert(msg['type'] == 'answer') print ('Received answer:\n%s' % sdp) res, sdpmsg = GstSdp.SDPMessage.new() GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg) answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg) promise = Gst.Promise.new() self.webrtc.emit('set-remote-description', answer, promise) promise.interrupt() # Extract ICE candidates from the SDP to work around a GStreamer # limitation in (at least) 1.16.2 and below self.extract_ice_from_sdp (sdp) elif 'ice' in msg: ice = msg['ice'] candidate = ice['candidate'] sdpmlineindex = ice['sdpMLineIndex'] self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate) async def loop(self): signaling = self.signaling await signaling.connect() await signaling.attach("janus.plugin.videoroom") loop = asyncio.get_event_loop() loop.create_task(signaling.keepalive()) #asyncio.create_task(self.keepalive()) joinmessage = { "request": "join", "ptype": "publisher", "room": 1234, "display": self.peer_id } await signaling.sendmessage(joinmessage) assert signaling.conn self.start_pipeline() while True: try: msg = await signaling.recv() if isinstance(msg, PluginData): if msg.jsep is not None: await self.handle_sdp(msg.jsep) elif isinstance(msg, Media): print (msg) elif isinstance(msg, WebrtcUp): print (msg) elif isinstance(msg, SlowLink): print (msg) elif isinstance(msg, HangUp): print (msg) elif not isinstance(msg, Ack): if 'candidate' in msg: ice = msg['candidate'] print (ice) if 'candidate' in ice: candidate = ice['candidate'] sdpmlineindex = ice['sdpMLineIndex'] self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate) print(msg) except (KeyboardInterrupt, ConnectionClosed): return return 0 async def close(self): return await self.signaling.close() def check_plugins(): needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp", "rtpmanager", "videotestsrc", "audiotestsrc"] missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed)) if len(missing): print('Missing gstreamer plugins:', missing) return False return True if __name__=='__main__': Gst.init(None) if not check_plugins(): sys.exit(1) parser = argparse.ArgumentParser() parser.add_argument('label', help='videoroom label') parser.add_argument('--server', help='Signalling server to connect to, eg "wss://127.0.0.1:8989"') args = parser.parse_args() c = WebRTCClient(args.label, args.server) loop = asyncio.get_event_loop() try: loop.run_until_complete( c.loop() ) except KeyboardInterrupt: pass finally: print("Interrupted, cleaning up") loop.run_until_complete(c.close())