#!/usr/bin/python3 import argparse import asyncio import json import logging import ssl import queue from typing import Optional, List import gi import websockets 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 gi.require_version('GstRtspServer', '1.0') from gi.repository import Gst, GstRtspServer, GObject, GLib log = logging.getLogger(__name__) class RtspServer: def __init__(self): server = GstRtspServer.RTSPServer() server.set_address("::") server.set_service('8554') # port as string factory = GstRtspServer.RTSPMediaFactory() # factory.set_launch("intervideosrc ! decodebin ! theoraenc ! queue ! rtptheorapay name=pay0") # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! theoraenc ! queue ! rtptheorapay name=pay0") # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,format=I420,framerate=10/1 ! theoraenc ! queue ! rtptheorapay name=pay0") # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc ! queue ! rtph264pay pt=96 name=pay0") factory.set_launch("intervideosrc ! decodebin ! x264enc ! queue ! rtph264pay pt=96 name=pay0") # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,framerate=10/1 ! x264enc ! queue ! rtph264pay pt=96 name=pay0") factory.set_shared(True) mountPoints = server.get_mount_points() mountPoints.add_factory("/cug", factory) server.attach() self.server = server class Events: def __init__(self): self.sdp_offer = queue.Queue() self.sdp_answer = queue.Queue() self.generated_ice_candidates = queue.Queue() self.received_ice_candidates = queue.Queue() self.sdp_info = queue.Queue() # (sdp_mids, user_fragments) self.room_left = queue.Queue() class SignalingClient: def __init__(self, events: Events, uri): self.events = events self.uri = uri self.ssl_context = ssl.SSLContext() self.ssl_context.check_hostname = False self.ssl_context.verify_mode = ssl.CERT_NONE self.session_id = None async def receive(self, uri): async for msg in self.websocket: msg_json = json.loads(msg) msg_type = msg_json['Type'] msg_value = msg_json['Value'] assert self.session_id is None or self.session_id == msg_json['SessionID'] if msg_type == 'newSession': self.session_id = msg_json['SessionID'] log.info(f"New session {self.session_id}") elif msg_type == 'gotOffer': value_json = json.loads(msg_value) sdp = value_json['sdp'] log.info(f'Got SDP offer') log.debug(f'SDP offer:\n{sdp}') self.events.sdp_offer.put_nowait(sdp) elif msg_type == 'addCallerIceCandidate': value_json = json.loads(msg_value) log.info(f'Got ICE candidate') log.debug(f'ICE candidate: {value_json}') self.events.received_ice_candidates.put_nowait(value_json) elif msg_type == 'roomNotFound': log.error(f'The room was not found: {uri}') return elif msg_type == 'roomClosed': log.info(f'Oh noes, the room went away (session {self.session_id})!') self.events.room_left.put_nowait(True) return else: log.error(f'Unknown message type {msg_type}') async def send(self): sdp_mids = None user_fragments = None while True: if self.events.sdp_answer.qsize() > 0: sdp_answer = self.events.sdp_answer.get_nowait() sdp_answer_msg = json.dumps({ 'SessionID': self.session_id, 'Type': "gotAnswer", 'Value': json.dumps({ 'type': 'answer', 'sdp': sdp_answer }) }) await self.websocket.send(sdp_answer_msg) elif self.events.sdp_info.qsize() > 0: sdp_mids, user_fragments = self.events.sdp_info.get_nowait() elif self.events.generated_ice_candidates.qsize() > 0 \ and sdp_mids is not None and user_fragments is not None: mlineindex, candidate = self.events.generated_ice_candidates.get_nowait() sdp_mid = sdp_mids[mlineindex] user_fragment = user_fragments[mlineindex] icemsg_value = json.dumps({ "candidate": candidate, "sdpMid": sdp_mid, "sdpMLineIndex": mlineindex, "usernameFragment": user_fragment, }) icemsg = json.dumps({ 'SessionID': self.session_id, 'Type': 'addCalleeIceCandidate', 'Value': icemsg_value, }) log.info(f'Send ICE candidate') log.debug(f'ICE candidate: {icemsg_value}') await self.websocket.send(icemsg) else: await asyncio.sleep(0.2) async def run(self): self.session_id = None async with websockets.connect(self.uri, ssl=self.ssl_context, close_timeout=0.5) as self.websocket: receive_task = asyncio.Task(self.receive(self.uri)) send_task = asyncio.Task(self.send()) done, pending = await asyncio.wait([receive_task, send_task], return_when=asyncio.FIRST_COMPLETED) for task in pending: task.cancel() class WebRTCClient: def __init__(self, events: Events): self.events = events self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace') self.pipe = Gst.Pipeline.new("pipeline") Gst.Bin.do_add_element(self.pipe, self.webrtcbin) self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed) self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate) self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added) def on_negotiation_needed(self, element): log.info('on_negotiation_needed') def on_ice_candidate(self, element, mlineindex, candidate): log.info('on_ice_candidate') self.events.generated_ice_candidates.put_nowait((mlineindex, candidate)) def webrtcbin_pad_added(self, element, pad): log.info('webrtcbin_pad_added') if pad.direction != Gst.PadDirection.SRC: return decodebin = Gst.ElementFactory.make('decodebin') decodebin.connect('pad-added', self.decodebin_pad_added) self.pipe.add(decodebin) decodebin.sync_state_with_parent() self.webrtcbin.link(decodebin) def decodebin_pad_added(self, element, pad): log.info('decodebin_pad_added') if not pad.has_current_caps(): log.info(pad, 'has no caps, ignoring') return caps = pad.get_current_caps() padsize = caps.get_size() log.info(f'>>>> {padsize} {caps}') for i in range(padsize): s = caps.get_structure(i) # Gst.Structure name = s.get_name() log.info(f'###### {name}') if name.startswith('video'): q = Gst.ElementFactory.make('queue') conv = Gst.ElementFactory.make('videoconvert') enc = Gst.ElementFactory.make('x264enc') enc.set_property('bitrate', 1000) enc.set_property('tune', 'zerolatency') capsfilter = Gst.ElementFactory.make('capsfilter') capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc')) flmux = Gst.ElementFactory.make('flvmux') sink = Gst.ElementFactory.make('rtmpsink') sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa') # sink.set_property('location', 'rtmp://bla:1936/gregoa') print(sink.props.location, dir(sink.props)) assert q and conv and enc and capsfilter and flmux and sink self.pipe.add(q) self.pipe.add(conv) self.pipe.add(enc) self.pipe.add(capsfilter) self.pipe.add(flmux) self.pipe.add(sink) q_pad_sink = q.get_static_pad('sink') assert q_pad_sink pad_link_return = pad.link(q_pad_sink) assert pad_link_return == Gst.PadLinkReturn.OK # ok = element.link(q) # assert ok ok = q.link(conv) assert ok ok = conv.link(enc) assert ok ok = enc.link(capsfilter) assert ok ok = capsfilter.link(flmux) assert ok ok = flmux.link(sink) assert ok self.pipe.set_state(Gst.State.PLAYING) self.pipe.sync_children_states() #print(dir(Gst.DebugGraphDetails)) #Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL) 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 set_remote_desciption_done(self, gst_promise): gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done) self.webrtcbin.emit('create-answer', None, gst_promise) def create_answer_done(self, gst_promise): reply = gst_promise.get_reply() answer = reply.get_value('answer') sdp_message = answer.sdp mids = [sdp_message.get_media(i).get_attribute_val('mid') for i in range(sdp_message.medias_len())] user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag') for i in range(sdp_message.medias_len())] self.events.sdp_info.put_nowait((mids, user_fragments)) sdp_answer = sdp_message.as_text() log.info(f'Send SDP answer') log.debug(f'SDP answer:\n{sdp_answer}') self.events.sdp_answer.put_nowait(sdp_answer) gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done) self.webrtcbin.emit('set-local-description', answer, gst_promise) def set_local_description_done(self, gst_promise): gst_promise.get_reply() async def run(self): bus = Gst.Pipeline.get_bus(self.pipe) self.pipe.set_state(Gst.State.PLAYING) try: while True: if bus.have_pending(): msg = bus.pop() if msg.type == Gst.MessageType.ERROR: log.error(f'Error from gstreamer message bus: {msg.get_structure()}') return elif msg.type == Gst.MessageType.EOS: # end of stream log.info(f'Gstreamer message bus reports end of stream') return elif self.events.sdp_offer.qsize() > 0: sdp_offer = self.events.sdp_offer.get_nowait() res, sm = GstSdp.SDPMessage.new() assert res == GstSdp.SDPResult.OK GstSdp.sdp_message_parse_buffer(bytes(sdp_offer.encode()), sm) # the three lines above can also be done this way in new versions of GStreamer: # sm = GstSdp.SDPMessage.new_from_text(sdp_offer) rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm) gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done) self.webrtcbin.emit('set-remote-description', rd, gst_promise) elif self.events.received_ice_candidates.qsize() > 0: ic = self.events.received_ice_candidates.get_nowait() if ic['candidate'] != '': self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate']) elif self.events.room_left.qsize() > 0: self.events.room_left.get_nowait() return else: await asyncio.sleep(0.1) finally: self.pipe.set_state(Gst.State.NULL) async def run_repeated(task): while True: await task() await asyncio.sleep(0.1) async def run(uri): try: events = Events() # rtsp = RtspServer() webrtc = WebRTCClient(events) signaling = SignalingClient(events, uri) webrtc_task = asyncio.Task(webrtc.run()) signaling_task = asyncio.Task(signaling.run()) done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED) for task in done: task.result() for task in pending: task.cancel() except OSError as e: print(e) def main(): logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s') parser = argparse.ArgumentParser() parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug', help='Signalling server URI') args = parser.parse_args() Gst.init(None) asyncio.run(run(args.uri), debug=True) if __name__ == '__main__': main()