def setReceiver(self, offer_sdp):
'''
'''
print('setReceiver thread:', threading.get_ident())
result, offer = GstSdp.SDPMessage.new()
assert (result == GstSdp.SDPResult.OK)
GstSdp.sdp_message_parse_buffer(offer_sdp.encode(), offer)
description = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.OFFER, offer)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', description, promise)
promise.interrupt()
direction_audio = GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV
caps_audio = Gst.caps_from_string(
"application/x-rtp,media=audio,encoding-name=OPUS,payload=111")
direction_video = GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV
caps_vedio = Gst.caps_from_string(
"application/x-rtp,media=video,encoding-name=VP8,payload=96")
self.webrtc.emit('add-transceiver', direction_audio, caps_audio)
self.webrtc.emit('add-transceiver', direction_video, caps_vedio)
promise = Gst.Promise.new()
self.webrtc.emit('create-answer', None, promise)
promise.wait()
reply = promise.get_reply()
answer = reply.get_value('answer')
# promise = Gst.Promise.new_with_change_func(self.onSetLocalDescription, 'myaatga')
promise = Gst.Promise.new()
self.webrtc.emit('set-local-description', answer, promise)
promise.wait()
return answer.sdp.as_text()
async def handle_sdp(self, message):
if self.state != 'publish':
return
assert (self.webrtc)
if message['message'] == 'setRemoteSDP':
sdp = message['data'][1]
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
# Flashphoner returns a sdp that doesn't have the "setup" attribute
# Which Gstreamer rejects.
# So we add it back
for i in range(N_MEDIA):
sdpmedia = sdpmsg.get_media(i)
if not sdpmedia.get_attribute_val("setup") in [
"actpass", "active", "passive"
]:
print("Setting 'setup' attribute for media %d" % i)
sdpmedia.add_attribute("setup", "active")
answer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', answer, promise)
promise.interrupt()
for i in range(N_MEDIA):
sdpmedia = sdpmsg.get_media(i)
candidate = sdpmedia.get_attribute_val_n('candidate', 0)
self.webrtc.emit('add-ice-candidate', i,
'a=candidate:' + candidate)
def msg_handler(self, msg):
#print("receiver", msg)
if msg.startswith("ROOM_OK"):
self.em.emit("change_label",
label_id="receiver_room_id_label",
new_text=self.room_id)
self.em.emit("remove_container",
container_id="receiver_connect_container",
window="receiver")
elif msg.startswith('ROOM_PEER_MSG'):
data = json.loads(msg.split(maxsplit=2)[2])
if "sdp" in data:
sdp = data["sdp"]
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
offer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new_with_change_func(
self.on_remote_dec_set, None)
self.webrtcbin.emit("set-remote-description", offer, promise)
elif "ice" in data:
mline_index = int(data["ice"]["sdpMLineIndex"])
candidate = data["ice"]["candidate"]
self.webrtcbin.emit("add-ice-candidate", mline_index,
candidate)
def handle_sdp(self, message):
assert (self.webrtc)
msg = json.loads(message)
if 'sdp' in msg:
sdp = msg['sdp']
print(str(sdp))
if (sdp['type'] == 'answer'):
time.sleep(2)
sdp = sdp['sdp']
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()
# loop = asyncio.get_event_loop()
# loop.run_until_complete(self.send_data())
elif (sdp['type'] == 'offer'):
# peer = sdp['peer_id']
sdp = sdp['sdp']
peer = 2
print('Received offer:\n%s' % sdp)
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
def set_sdp(self, sdp_type, sdp):
"""Sets remote SDP received by peer.
Arguments:
sdp_type {string} -- type of sdp, offer or answer
sdp {object} -- SDP object
Raises:
GSTWebRTCAppError -- thrown if SDP is recevied before session has been started.
GSTWebRTCAppError -- thrown if SDP type is not 'answer', this script initiates the call, not the peer.
"""
if not self.webrtcbin:
raise GSTWebRTCAppError('Received SDP before session started')
if sdp_type != 'answer':
raise GSTWebRTCAppError('ERROR: sdp type was not "answer"')
_, 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.webrtcbin.emit('set-remote-description', answer, promise)
promise.interrupt()
async def connection_handler(self, ws):
print("Connection attempt")
self.current_ws = ws
self.start_pipeline()
await ws.send(self.sdp_queue.get())
while True:
await ws.send(self.ice_queue.get())
msg = await ws.recv()
msg = json.loads(msg)
print(msg)
if msg['type'] == 'sdp':
sdp = msg['data']
sdp = sdp['sdp']
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()
if msg['type'] == 'ice':
print('Received ICE:\n%s' % sdp)
ice = msg['data']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
else:
print('Unsupported message type')
async def on_message(self, conn, path):
while True:
data = await conn.recv();
print(f'{data}')
datas = data.split(' ')
if datas[0] == 'HELLO':
await conn.send('HELLO')
elif datas[0] == 'SESSION':
await conn.send('SESSION_OK')
else:
msg = json.loads(data)
if 'sdp' in msg:
sdp = msg['sdp']
assert(sdp['type'] == 'offer')
sdp = sdp['sdp']
sdp = 'v=0\no=- 249052402997811464 2 IN IP4 127.0.0.1\ns=-\nt=0 0\na=group:BUNDLE 0 1\na=extmap-allow-mixed\na=msid-semantic: WMS\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\nc=IN IP4 0.0.0.0\na=rtcp:9 IN IP4 0.0.0.0\na=ice-ufrag:09iz\na=ice-pwd:TDNEa4DNQNu5vntI7paNKKgV\na=ice-options:trickle\na=fingerprint:sha-256 4F:12:50:51:FE:1A:76:0E:74:6D:79:31:DB:C0:E5:42:41:4F:AA:9D:AA:C9:29:AD:A2:49:6E:F3:AF:0F:3A:14\na=setup:actpass\na=mid:0\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\na=sendonly\na=rtcp-mux\na=rtpmap:111 opus/48000/2\na=rtcp-fb:111 transport-cc\na=fmtp:111 minptime=10;useinbandfec=1\na=rtpmap:103 ISAC/16000\na=rtpmap:104 ISAC/32000\na=rtpmap:9 G722/8000\na=rtpmap:0 PCMU/8000\na=rtpmap:8 PCMA/8000\na=rtpmap:106 CN/32000\na=rtpmap:105 CN/16000\na=rtpmap:13 CN/8000\na=rtpmap:110 telephone-event/48000\na=rtpmap:112 telephone-event/32000\na=rtpmap:113 telephone-event/16000\na=rtpmap:126 telephone-event/8000\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 122 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 35\nc=IN IP4 0.0.0.0\na=rtcp:9 IN IP4 0.0.0.0\na=ice-ufrag:09iz\na=ice-pwd:TDNEa4DNQNu5vntI7paNKKgV\na=ice-options:trickle\na=fingerprint:sha-256 4F:12:50:51:FE:1A:76:0E:74:6D:79:31:DB:C0:E5:42:41:4F:AA:9D:AA:C9:29:AD:A2:49:6E:F3:AF:0F:3A:14\na=setup:actpass\na=mid:1\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\na=extmap:13 urn:3gpp:video-orientation\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\na=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\na=sendonly\na=rtcp-mux\na=rtcp-rsize\na=rtpmap:96 VP8/90000\na=rtcp-fb:96 goog-remb\na=rtcp-fb:96 transport-cc\na=rtcp-fb:96 ccm fir\na=rtcp-fb:96 nack\na=rtcp-fb:96 nack pli\na=rtpmap:97 rtx/90000\na=fmtp:97 apt=96\na=rtpmap:98 VP9/90000\na=rtcp-fb:98 goog-remb\na=rtcp-fb:98 transport-cc\na=rtcp-fb:98 ccm fir\na=rtcp-fb:98 nack\na=rtcp-fb:98 nack pli\na=fmtp:98 profile-id=0\na=rtpmap:99 rtx/90000\na=fmtp:99 apt=98\na=rtpmap:100 VP9/90000\na=rtcp-fb:100 goog-remb\na=rtcp-fb:100 transport-cc\na=rtcp-fb:100 ccm fir\na=rtcp-fb:100 nack\na=rtcp-fb:100 nack pli\na=fmtp:100 profile-id=2\na=rtpmap:101 rtx/90000\na=fmtp:101 apt=100\na=rtpmap:122 VP9/90000\na=rtcp-fb:122 goog-remb\na=rtcp-fb:122 transport-cc\na=rtcp-fb:122 ccm fir\na=rtcp-fb:122 nack\na=rtcp-fb:122 nack pli\na=fmtp:122 profile-id=1\na=rtpmap:102 H264/90000\na=rtcp-fb:102 goog-remb\na=rtcp-fb:102 transport-cc\na=rtcp-fb:102 ccm fir\na=rtcp-fb:102 nack\na=rtcp-fb:102 nack pli\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\na=rtpmap:121 rtx/90000\na=fmtp:121 apt=102\na=rtpmap:127 H264/90000\na=rtcp-fb:127 goog-remb\na=rtcp-fb:127 transport-cc\na=rtcp-fb:127 ccm fir\na=rtcp-fb:127 nack\na=rtcp-fb:127 nack pli\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\na=rtpmap:120 rtx/90000\na=fmtp:120 apt=127\na=rtpmap:125 H264/90000\na=rtcp-fb:125 goog-remb\na=rtcp-fb:125 transport-cc\na=rtcp-fb:125 ccm fir\na=rtcp-fb:125 nack\na=rtcp-fb:125 nack pli\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\na=rtpmap:107 rtx/90000\na=fmtp:107 apt=125\na=rtpmap:108 H264/90000\na=rtcp-fb:108 goog-remb\na=rtcp-fb:108 transport-cc\na=rtcp-fb:108 ccm fir\na=rtcp-fb:108 nack\na=rtcp-fb:108 nack pli\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\na=rtpmap:109 rtx/90000\na=fmtp:109 apt=108\na=rtpmap:124 H264/90000\na=rtcp-fb:124 goog-remb\na=rtcp-fb:124 transport-cc\na=rtcp-fb:124 ccm fir\na=rtcp-fb:124 nack\na=rtcp-fb:124 nack pli\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\na=rtpmap:119 rtx/90000\na=fmtp:119 apt=124\na=rtpmap:123 H264/90000\na=rtcp-fb:123 goog-remb\na=rtcp-fb:123 transport-cc\na=rtcp-fb:123 ccm fir\na=rtcp-fb:123 nack\na=rtcp-fb:123 nack pli\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\na=rtpmap:118 rtx/90000\na=fmtp:118 apt=123\na=rtpmap:114 red/90000\na=rtpmap:115 rtx/90000\na=fmtp:115 apt=114\na=rtpmap:116 ulpfec/90000\na=rtpmap:35 flexfec-03/90000\na=rtcp-fb:35 goog-remb\na=rtcp-fb:35 transport-cc\na=fmtp:35 repair-window=10000000\n'
self.start_pipeline(conn)
print('set-remote-description')
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', offer, promise)
promise.interrupt()
promise = Gst.Promise.new_with_change_func(self.on_answer_created, conn, None)
self.webrtc.emit('create-answer', None, promise)
elif 'ice' in msg:
print('[RECV] ice:', msg)
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
def handle_sdp(self, message):
msg = json.loads(message)
if 'sdp' in msg:
sdp = msg['sdp']
sdp_type_str = sdp['type']
sdp_msg_str = sdp['sdp']
#parse sdp['sdp'] back to a GstSdp.SDPMessage
_res, sdp_msg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp_msg_str.encode()), sdp_msg)
#parse sdp['type'] back to a GstWebRTC.WebRTCSDPType
sdp_type = parse_WebRTCSDPType(sdp_type_str)
if sdp_type == None:
self.node.get_logger().error('invalid sdp type')
return
# combine the parsed sdp into a GstWebRTC.WebRTCSessionDescription
remote_sdp = GstWebRTC.WebRTCSessionDescription.new(sdp_type, sdp_msg)
if self.remote_sends_sdp_cb != None:
self.remote_sends_sdp_cb(remote_sdp)
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
if self.remote_sends_ice_cb != None:
self.remote_sends_ice_cb(sdpmlineindex, candidate)
def start_pipeline(self):
self.pipe = Gst.parse_launch(PIPELINE_DESC)
self.webrtc = self.pipe.get_by_name('sendrecv')
self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
self.webrtc.connect('pad-added', self.on_incoming_stream)
self.pipe.set_state(Gst.State.PLAYING)
offer = 'v=0\no=- 249052402997811464 2 IN IP4 127.0.0.1\ns=-\nt=0 0\na=group:BUNDLE 0 1\na=extmap-allow-mixed\na=msid-semantic: WMS\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\nc=IN IP4 0.0.0.0\na=rtcp:9 IN IP4 0.0.0.0\na=ice-ufrag:09iz\na=ice-pwd:TDNEa4DNQNu5vntI7paNKKgV\na=ice-options:trickle\na=fingerprint:sha-256 4F:12:50:51:FE:1A:76:0E:74:6D:79:31:DB:C0:E5:42:41:4F:AA:9D:AA:C9:29:AD:A2:49:6E:F3:AF:0F:3A:14\na=setup:actpass\na=mid:0\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\na=sendrecv\na=rtcp-mux\na=rtpmap:111 opus/48000/2\na=rtcp-fb:111 transport-cc\na=fmtp:111 minptime=10;useinbandfec=1\na=rtpmap:103 ISAC/16000\na=rtpmap:104 ISAC/32000\na=rtpmap:9 G722/8000\na=rtpmap:0 PCMU/8000\na=rtpmap:8 PCMA/8000\na=rtpmap:106 CN/32000\na=rtpmap:105 CN/16000\na=rtpmap:13 CN/8000\na=rtpmap:110 telephone-event/48000\na=rtpmap:112 telephone-event/32000\na=rtpmap:113 telephone-event/16000\na=rtpmap:126 telephone-event/8000\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 122 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116 35\nc=IN IP4 0.0.0.0\na=rtcp:9 IN IP4 0.0.0.0\na=ice-ufrag:09iz\na=ice-pwd:TDNEa4DNQNu5vntI7paNKKgV\na=ice-options:trickle\na=fingerprint:sha-256 4F:12:50:51:FE:1A:76:0E:74:6D:79:31:DB:C0:E5:42:41:4F:AA:9D:AA:C9:29:AD:A2:49:6E:F3:AF:0F:3A:14\na=setup:actpass\na=mid:1\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\na=extmap:13 urn:3gpp:video-orientation\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\na=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\na=sendrecv\na=rtcp-mux\na=rtcp-rsize\na=rtpmap:96 VP8/90000\na=rtcp-fb:96 goog-remb\na=rtcp-fb:96 transport-cc\na=rtcp-fb:96 ccm fir\na=rtcp-fb:96 nack\na=rtcp-fb:96 nack pli\na=rtpmap:97 rtx/90000\na=fmtp:97 apt=96\na=rtpmap:98 VP9/90000\na=rtcp-fb:98 goog-remb\na=rtcp-fb:98 transport-cc\na=rtcp-fb:98 ccm fir\na=rtcp-fb:98 nack\na=rtcp-fb:98 nack pli\na=fmtp:98 profile-id=0\na=rtpmap:99 rtx/90000\na=fmtp:99 apt=98\na=rtpmap:100 VP9/90000\na=rtcp-fb:100 goog-remb\na=rtcp-fb:100 transport-cc\na=rtcp-fb:100 ccm fir\na=rtcp-fb:100 nack\na=rtcp-fb:100 nack pli\na=fmtp:100 profile-id=2\na=rtpmap:101 rtx/90000\na=fmtp:101 apt=100\na=rtpmap:122 VP9/90000\na=rtcp-fb:122 goog-remb\na=rtcp-fb:122 transport-cc\na=rtcp-fb:122 ccm fir\na=rtcp-fb:122 nack\na=rtcp-fb:122 nack pli\na=fmtp:122 profile-id=1\na=rtpmap:102 H264/90000\na=rtcp-fb:102 goog-remb\na=rtcp-fb:102 transport-cc\na=rtcp-fb:102 ccm fir\na=rtcp-fb:102 nack\na=rtcp-fb:102 nack pli\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\na=rtpmap:121 rtx/90000\na=fmtp:121 apt=102\na=rtpmap:127 H264/90000\na=rtcp-fb:127 goog-remb\na=rtcp-fb:127 transport-cc\na=rtcp-fb:127 ccm fir\na=rtcp-fb:127 nack\na=rtcp-fb:127 nack pli\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\na=rtpmap:120 rtx/90000\na=fmtp:120 apt=127\na=rtpmap:125 H264/90000\na=rtcp-fb:125 goog-remb\na=rtcp-fb:125 transport-cc\na=rtcp-fb:125 ccm fir\na=rtcp-fb:125 nack\na=rtcp-fb:125 nack pli\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\na=rtpmap:107 rtx/90000\na=fmtp:107 apt=125\na=rtpmap:108 H264/90000\na=rtcp-fb:108 goog-remb\na=rtcp-fb:108 transport-cc\na=rtcp-fb:108 ccm fir\na=rtcp-fb:108 nack\na=rtcp-fb:108 nack pli\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\na=rtpmap:109 rtx/90000\na=fmtp:109 apt=108\na=rtpmap:124 H264/90000\na=rtcp-fb:124 goog-remb\na=rtcp-fb:124 transport-cc\na=rtcp-fb:124 ccm fir\na=rtcp-fb:124 nack\na=rtcp-fb:124 nack pli\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f\na=rtpmap:119 rtx/90000\na=fmtp:119 apt=124\na=rtpmap:123 H264/90000\na=rtcp-fb:123 goog-remb\na=rtcp-fb:123 transport-cc\na=rtcp-fb:123 ccm fir\na=rtcp-fb:123 nack\na=rtcp-fb:123 nack pli\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f\na=rtpmap:118 rtx/90000\na=fmtp:118 apt=123\na=rtpmap:114 red/90000\na=rtpmap:115 rtx/90000\na=fmtp:115 apt=114\na=rtpmap:116 ulpfec/90000\na=rtpmap:35 flexfec-03/90000\na=rtcp-fb:35 goog-remb\na=rtcp-fb:35 transport-cc\na=fmtp:35 repair-window=10000000\n'
print('offer sdp : ' , offer)
sdp = offer
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', offer, promise)
promise.interrupt()
direction_a = GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY
caps_a = Gst.caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,payload=111")
direction_v = GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY
caps_v = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8,payload=96")
self.webrtc.emit('add-transceiver', direction_a, caps_a)
self.webrtc.emit('add-transceiver', direction_v, caps_v)
promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
self.webrtc.emit('create-answer', None, promise)
async def signaling(request, ws):
logging.debug("is called!")
clients.append(id(ws))
self.websocket = ws
while True:
#pprint(ws)
#print("websocket id: {id}".format(id=id(ws)))
# a Python object (dict):
data = {"name": "John"}
logging.debug('Sending: ' + str(data))
await ws.send(json.dumps(data))
message = await ws.recv()
logging.debug('Received: ' + message)
# https://realpython.com/python-json/
msg = json.loads(message)
if msg.get('join'):
self.rtcClient.create_offer()
if msg.get('answer'):
sdp = msg['answer']
_, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()),
sdpmsg)
answer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
self.rtcClient.set_remote_description(answer)
if msg.get('candidate') and msg['candidate'].get('candidate'):
print('add_ice_candidate')
self.rtcClient.add_ice_candidate(msg['candidate'])
def _deepcopy_session_description(self, desc):
# XXX: passing 'offer' to both a promise and an action signal without
# a deepcopy will segfault...
new_sdp = GstSdp.SDPMessage.new()[1]
GstSdp.sdp_message_parse_buffer(bytes(desc.sdp.as_text().encode()),
new_sdp)
return GstWebRTC.WebRTCSessionDescription.new(desc.type, new_sdp)
def take_configuration(self, data):
assert (self.webrtc)
assert(data['type'] == 'answer')
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(data['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()
def setRemoteDescription(self, desc):
sdp = desc['sdp']
t = GstWebRTC.WebRTCSDPType.ANSWER
if (desc['type'] == 'offer'):
t = GstWebRTC.WebRTCSDPType.OFFER
_, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
answer = GstWebRTC.WebRTCSessionDescription.new(t, sdpmsg)
promise = Gst.Promise.new_with_change_func(self.onRemoteAnswerSet,
None)
self.webrtc.emit('set-remote-description', answer, promise)
def setAnswer(self, 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()
print('set-remote-description end..')
async def sdp_message_from_peer(self, ws, sdp):
'''
Called when the peer (client) has sent (via websocket) an SDP message
'''
assert(sdp['type'] == 'answer')
sdp = sdp['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.peers[ws]['webrtcbin'].emit('set-remote-description', answer, promise)
promise.interrupt()
async def handle_sdp(self, sdp):
assert self.webrtc
assert (sdp['type'] == 'answer')
sdp = sdp['sdp']
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()
def set_remote_sdp(self, sdp: str):
"""Set remote session description.
:param sdp: Session description
:type sdp: str
"""
log.info("Setting remote session description")
_, sdp_msg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdp_msg)
answer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.ANSWER, sdp_msg)
promise = Gst.Promise.new()
self.webrtc.emit("set-remote-description", answer, promise)
promise.interrupt()
def have_json(msg):
if 'sdp' in msg:
sdp = msg['sdp']
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp['sdp'].encode()),
sdpmsg)
sdptype = GstWebRTC.WebRTCSDPType.ANSWER if sdp[
'type'] == 'answer' else GstWebRTC.WebRTCSDPType.OFFER
desc = GstWebRTC.WebRTCSessionDescription.new(sdptype, sdpmsg)
self.client.set_remote_description(desc)
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.client.add_ice_candidate(sdpmlineindex, candidate)
def handle_offer(self, msg, UUID):
print("HANDLE SDP OFFER")
client = self.clients[UUID]
if not client or not client['webrtc']:
return
if 'sdp' in msg:
print("INCOMDING OFFER SDP TYPE: "+msg['type']);
assert(msg['type'] == 'offer')
sdp = msg['sdp']
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new()
client['webrtc'].emit('set-remote-description', offer, promise)
promise.interrupt()
promise2 = Gst.Promise.new_with_change_func(self.on_answer_created, client['webrtc'], client)
client['webrtc'].emit('create-answer', None, promise2)
else:
print("No SDP as expected")
def handle_sdp(self, msg):
assert (self.webrtc)
res, sdpmsg = GstSdp.SDPMessage.new()
result = GstSdp.sdp_message_parse_buffer(bytes(msg.encode()), sdpmsg)
print("handle_sdp: ", result, sdpmsg, msg)
answer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', answer, promise)
promise.interrupt()
def handle_sdp(webrtc, message):
msg = json.loads(message)
if "sdp" in msg:
sdp = msg["sdp"]
assert sdp["type"] == "answer"
sdp = sdp["sdp"]
logger.info("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()
webrtc.emit("set-remote-description", answer, promise)
promise.interrupt()
elif "ice" in msg:
ice = msg["ice"]
candidate = ice["candidate"]
sdpmlineindex = ice["sdpMLineIndex"]
webrtc.emit("add-ice-candidate", sdpmlineindex, candidate)
def handle_sdp(self, msg):
if 'sdpAnswer' in msg:
sdp = msg['sdpAnswer']
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()
elif 'candidate' in msg:
ice = msg['candidate']
candidate = ice['candidate']
if not self.check_ice_candidate(candidate):
return
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
def handle_sdp(self, message):
assert (self.webrtc)
msg = json.loads(message)
if 'sdp' in msg:
sdp = msg['sdp']
assert(sdp['type'] == 'answer')
sdp = sdp['sdp']
print ('Received answer, whee:\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()
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
def handle_sdp(self, message):
msg = json.loads(message)
if 'sdpAnswer' in msg:
sdp = msg['sdpAnswer']
print('received sdp answer')
print(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()
elif 'candidate' in msg:
ice = msg['candidate']
candidate = ice['candidate']
#print("got ice candidate: %s" % candidate)
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
else:
print(msg)
async def handle_peer_msg(self, message, peer_id):
assert (self.webrtc[peer_id])
msg = json.loads(message)
if 'sdp' in msg:
sdp = msg['sdp']
assert(sdp['type'] == 'answer')
sdp = sdp['sdp']
with open('webrtc.log', 'a+') as log:
log.write('Received answer:\n%s\n' % 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[peer_id].emit('set-remote-description', answer, promise)
promise.interrupt()
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc[peer_id].emit('add-ice-candidate', sdpmlineindex, candidate)
def handle_sdp_ice(self, msg, UUID):
client = self.clients[UUID]
if not client or not client['webrtc']:
return
if 'sdp' in msg:
print("INCOMING ANSWER SDP TYPE: "+msg['type'])
assert(msg['type'] == 'answer')
sdp = msg['sdp']
# 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()
client['webrtc'].emit('set-remote-description', answer, promise)
promise.interrupt()
elif 'candidate' in msg:
print("HANDLE ICE")
candidate = msg['candidate']
sdpmlineindex = msg['sdpMLineIndex']
client['webrtc'].emit('add-ice-candidate', sdpmlineindex, candidate)
else:
print("UNEXPECTED INCOMING")
def setRemoteDescription(self, offer_sdp):
'''
'''
print('setRemoteDescription thread:', threading.get_ident())
result, offer = GstSdp.SDPMessage.new()
assert (result == GstSdp.SDPResult.OK)
GstSdp.sdp_message_parse_buffer(offer_sdp.encode(), offer)
description = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.OFFER, offer)
self.webrtc.emit('set-remote-description', description, None)
promise = Gst.Promise.new()
self.webrtc.emit('create-answer', None, promise)
promise.wait()
reply = promise.get_reply()
answer = reply.get_value('answer')
# promise = Gst.Promise.new_with_change_func(self.onSetLocalDescription, 'myaatga')
promise = Gst.Promise.new()
self.webrtc.emit('set-local-description', answer, promise)
promise.wait()
return answer.sdp.as_text()
async def on_message(self, conn, path):
while True:
data = await conn.recv()
print(f'{data}')
datas = data.split(' ')
if datas[0] == 'HELLO':
await conn.send('HELLO')
elif datas[0] == 'SESSION':
await conn.send('SESSION_OK')
else:
msg = json.loads(data)
if 'sdp' in msg:
sdp = msg['sdp']
assert (sdp['type'] == 'offer')
sdp = sdp['sdp']
self.start_pipeline(conn)
print('set-remote-description')
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()),
sdpmsg)
offer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', offer, promise)
promise.interrupt()
promise = Gst.Promise.new_with_change_func(
self.on_answer_created, conn, None)
self.webrtc.emit('create-answer', None, promise)
elif 'ice' in msg:
print('[RECV] ice:', msg)
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex,
candidate)
async def handle_message(self, message):
#assert (self.webrtc)
msg = json.loads(message)
if 'webrtc' in msg:
video_formats = msg.get('webrtc_video')
audio_only = not video_formats
self.launch_x11vnc(audio_only)
time.sleep(1)
print('sending ice credentials')
await self.send_ice_credentials()
self.start_pipeline(video_formats, audio_only)
if 'ms_audio' in msg:
ms_audio = msg['ms_audio']
if ms_audio == 'reset':
self.sending = False
else:
await self.start_ms_audio(ms_audio)
if 'sdp' in msg:
sdp = msg['sdp']
assert (sdp['type'] == 'answer')
sdp = sdp['sdp']
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()
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
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)