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Forked from saket424/janusvideoroom.py
Created August 7, 2021 18:07
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to publish to videoroom 1234 as user gstwebrtcdemo use 'python3 janusvideoroom.py --server wss://janus.example.com:8989 gstwebrtcdemo'
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
@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
PIPELINE_DESC = '''
webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302 videotestsrc pattern=ball ! video/x-raw,width=320,height=240 ! videoconvert ! queue ! x264enc ! rtph264pay !
queue ! application/x-rtp,media=video,encoding-name=H264,payload=97 ! sendrecv.
'''
#PIPELINE_DESC = '''
#webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302 rtspsrc location=rtsp://user2:passwd2@192.168.155.166:88/videoMain ! queue ! application/x-rtp, media=video, encoding-name=H264, payload=96 ! rtph264depay ! queue ! rtph264pay config-interval=1 ! application/x-rtp, media=video, encoding-name=H264, payload=97 ! sendrecv.
#'''
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)
async def connect(self):
self.conn = await websockets.connect(self.server, subprotocols=['janus-protocol'], ssl=ssl.SSLContext())
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):
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, id_, peer_id, server, signaling):
self.id_ = id_
self.conn = None
self.pipe = None
self.webrtc = None
self.peer_id = peer_id
self.server = server or 'wss://127.0.0.1:8989'
self.signaling = signaling
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, 'sdpMid': "0"}
self.webrtc.emit('add-ice-candidate', mlineindex, candidate)
print (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)
self.pipe.set_state(Gst.State.PLAYING)
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
for line in sdp.splitlines():
if line.startswith("a=candidate"):
candidate = line[2:]
print ('Received remote ice-candidate : %s\n' % candidate)
self.webrtc.emit('add-ice-candidate', 0, candidate)
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):
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
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()
our_id = random.randrange(10, 10000)
signaling = JanusGateway(args.server)
c = WebRTCClient(our_id, args.label, args.server, signaling)
loop = asyncio.get_event_loop()
try:
loop.run_until_complete(
c.loop(signaling)
)
except KeyboardInterrupt:
pass
finally:
print("Interrupted, cleaning up")
loop.run_until_complete(signaling.close())
websockets
gobject
PyGObject
attrs
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