ice_port: Client has expired #1259

toon4003 · 2023-05-31T07:13:50Z

toon4003
May 31, 2023

I'm new to webrtc, I'm pushing rtsp stream to ovenmediaengine (Docker version) on my local network and want to pull the stream through ovenmediaengine by webrtc and websockets but no data is coming. However, I test demo link (http) the same uri(websocket uri) and everything is fine.
what did I miss? or What I need to do next?

Example logs

[2023-05-31 04:26:00.364] I [DQICETmout:15] Monitor | stream_metrics.cpp:136  | A session has been stopped 
playing #default#app/stream on the WebRTC publisher. Concurrent Viewers[WebRTC(0)/Stream total(0)/App total(0)]
[2023-05-31 04:26:00.364] I [DQICETmout:15] Signalling | rtc_signalling_server.cpp:450  | Client is disconnected: HttpConnection(0x7fa50d5b9410) : WebSocket <ClientSocket: 0x7fa509400210, #54, Connected, TCP, Nonblocking, 192.168.1.40:48482> TLS(Disabled) (#default#app / stream, ufrag: local: TyHD29, remote: qIC8)
[2023-05-31 04:26:08.602] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(16667.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:26:15.552] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(8333.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:27:12.474] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(16667.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:27:20.051] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(8333.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:28:05.029] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(16667.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:28:15.246] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(8333.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:29:21.687] I [SPRtcSig-t3333:9] Signalling | rtc_signalling_server.cpp:316  | New client is connected: <ClientSocket: 0x7fa50d440e10, #54, Connected, TCP, Nonblocking, 192.168.1.33:51978>
[2023-05-31 04:29:24.674] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(16667.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:29:28.279] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(8333.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:29:30.074] I [SPICE-t3478:25] ICE | ice_port.cpp:566  | Turn client has connected : <ClientSocket: 0x7fa4fbe01410, #56, Connected, TCP, Nonblocking, 192.168.1.33:51979>
[2023-05-31 04:29:32.133] I [SPRtcSig-t3333:9] Monitor | stream_metrics.cpp:114  | A new session has started playing #default#app/stream on the WebRTC publisher. WebRTC(1)/Stream total(1)/App total(1)
[2023-05-31 04:29:32.133] I [SPRtcSig-t3333:9] ICE | ice_port.cpp:317  | Added session: 103 (ufrag: BsIOk1:tLMh)
[2023-05-31 04:29:57.743] W [AW-LLHLS6:81] FMP4 Packager | fmp4_storage.cpp:307  | LLHLS stream(#default#app/stream) is creating segments that are too large(16667.000000 ms). It seems that the keyframe interval is longer than the configured segment size.
[2023-05-31 04:30:02.388] W [DQICETmout:15] ICE | ice_port.cpp:480  | Client (session id: 103) has expired
[2023-05-31 04:30:02.388] I [DQICETmout:15] WebRTC Publisher | webrtc_publisher.cpp:645  | IcePort is disconnected. : (#default#app/stream/103) reason(5)
[2023-05-31 04:30:02.389] I [DQICETmout:15] WebRTC Publisher | webrtc_publisher.cpp:547  | Stop command received : #default#app/stream/103
[2023-05-31 04:30:02.390] I [DQICETmout:15] Monitor | stream_metrics.cpp:136  | A session has been stopped playing #default#app/stream on the WebRTC publisher. Concurrent Viewers[WebRTC(0)/Stream total(0)/App total(0)]
[2023-05-31 04:30:02.390] I [DQICETmout:15] Signalling | rtc_signalling_server.cpp:450  | Client is disconnected: HttpConnection(0x7fa50d5b9a10) : WebSocket <ClientSocket: 0x7fa50d440e10, #54, Connected, TCP, Nonblocking, 192.168.1.33:51978> TLS(Disabled) (#default#app / stream, ufrag: local: BsIOk1, remote: tLMh)

docker ovenmediaengine

UDP port is open

Python script:

async def test_connection_and_get_stream(data, websocket):
	
	ice_server_url = f"turn:{PUBLIC_IP_ADDRESS}:3478?transport=tcp"
	ice_server_username = data['iceServers'][0]['username']
	ice_server_credential = data['iceServers'][0]['credential']

	ice_server = RTCIceServer(urls=ice_server_url, username=ice_server_username, credential=ice_server_credential)
	pc = RTCPeerConnection(configuration=RTCConfiguration(iceServers=[ice_server]))
	
	pc.addTransceiver("video", direction="recvonly")

	recorder = MediaBlackhole()
	await recorder.start()

	@pc.on("datachannel") # only sctp can receive
	async def on_datachannel(channel):
		print(' ------------ datachannel ------------ ')
		print(f'show datachannel :{channel}')

		@channel.on("open")
		async def on_open():
			async def send_pings():
				while True:
					await channel_send(channel, "ping %d" % time.time())
					time.sleep(1)

			asyncio.ensure_future(send_pings())

		@channel.on("message")
		async def on_message(message):
			channel_log(channel, "<", message)

			if isinstance(message, str) and message.startswith("ping"):
				# reply
				await channel_send(channel, "pong" + message[4:])
				
	@pc.on("iceconnectionstatechange")
	async def on_iceconnectionstatechange():
		print(' ------------ iceconnectionstatechange ------------ ')
		print(f'show iceconnectionstatechange :{pc.iceConnectionState}')
		return pc
				
	@pc.on("icegatheringstatechange")
	async def on_icegatheringstatechange():
		print(' ------------ icegatheringstatechange ------------ ')
		print(f'show icegatheringstatechange :{pc.iceGatheringState}')
		
		return pc

	@pc.on("signalingstatechange")
	async def on_signalingstatechange():

		print(' ------------ signalingstatechange ------------ ')
		print(f'show signalingstatechange :{pc.signalingState}')

	@pc.on("track")
	async def on_track(track):
		"""
		Track has been received from client.
		"""
		pc.addTrack(MyVideoTrack())

		if track.kind == "audio":
			# pc.addTrack(self.player.audio)
			recorder.addTrack(track)

		elif track.kind == "video":
	
			while True:

				_, frame = await track.recv()
				
				if not frame:
					# print(f"Track {track.kind} not received done")
					break

				img = frame.to_ndarray(format="bgr24")

			recorder.addTrack(frame)

		@track.on('ended')
		async def on_track_ended():
			recorder.stop()

	@pc.on("stream")
	async def on_stream(stream):
		# Handle stream events
		print(' ------------ stream ------------ ')
		frame = await stream.recv()

	index = data['candidates'][0]['candidate'].find(':')
	candidate = data['candidates'][0]['candidate'][index+1:].split()

	ice_candidate = RTCIceCandidate(
		component=candidate[1],
		foundation=candidate[0],
		ip=PUBLIC_IP_ADDRESS,
		port=candidate[5],
		priority=candidate[3],
		protocol=candidate[2],
		type=candidate[-1],
		relatedAddress=PUBLIC_IP_ADDRESS)

	# sdp: type rtp
	if 'sdpMid' in list(data['candidates'][0].keys()):
		ice_candidate.sdpMid = data['candidates'][0]['spdMid']

	if 'sdpMLineIndex' in list(data['candidates'][0].keys()):
		ice_candidate.sdpMLineIndex = data['candidates'][0]['sdpMLineIndex']

	await pc.addIceCandidate(ice_candidate)

	offer_sdp_txt = data["sdp"]['sdp'] 
	offer_sdp = RTCSessionDescription(offer_sdp_txt, type=data["sdp"]['type'])
	await pc.setRemoteDescription(offer_sdp)
	
	# send answer
	answer = await pc.createAnswer()
	await pc.setLocalDescription(answer)

	while True:
		time.sleep(2)

		print("pc.iceGatheringState: %s" % pc.iceGatheringState)
		print("pc.connectionState : %s" % pc.connectionState) 

async def test_pull_stream():

	host = f'ws://{PUBLIC_IP_ADDRESS}:3333/app/stream?transport=tcp'
	print(f'connect to {host}...')

	async with websockets.connect(host) as websocket:

		if websocket.open:          

			await websocket.send(json.dumps({"command":"request_offer"}))
			answer_request = await websocket.recv()
			answer_request = json.loads(answer_request)

			if 'command' in answer_request and answer_request['command'] == "offer":

				# throw data to test_connection_and_get_stream()
				confirm_candidate = await test_connection_and_get_stream(answer_request, websocket)	
		
loop = asyncio.get_event_loop()
loop.run_until_complete(test_pull_stream())

Server.xml.txt
Here logs :ovenmediaengine.log

Answered by getroot

May 31, 2023

This is an issue where WebRTC communication fails after signaling is complete. Specifically, your WebRTC library is not sending or receiving Stun Binding Requests to OME. It's most likely an incorrect exchange of candidates.

Please refer to OvenRtcTester developed in GO language. This is very similar to yours.

https://github.com/AirenSoft/OvenMediaEngine/blob/master/misc/oven_rtc_tester/OvenRtcTester.go

View full answer

getroot · 2023-05-31T07:29:45Z

getroot
May 31, 2023
Maintainer Sponsor

	ice_server_url = f"turn:{PUBLIC_IP_ADDRESS}:3478?transport=tcp"
	ice_server_username = data['iceServers'][0]['username']
	ice_server_credential = data['iceServers'][0]['credential']

For ice_server_url, try to use the information given by the server like username and credential.

6 replies

toon4003 May 31, 2023
Author

sorry I miss understand, I change IP address like your mention but it's still the same.

getroot May 31, 2023
Maintainer Sponsor

I can't actually understand your code. But from my point of view, this is also the wrong part.

	ice_candidate = RTCIceCandidate(
		component=candidate[1],
		foundation=candidate[0],
		ip=PUBLIC_IP_ADDRESS,
		port=candidate[5],
		priority=candidate[3],
		protocol=candidate[2],
		type=candidate[-1],
		relatedAddress=PUBLIC_IP_ADDRESS)

You should provide OME's candidates received through signaling to the WebRTC library you are using, but you are providing your public IP instead. Since this is not an issue with OME, it would be better to contact the WebRTC library you are using for further assistance.

toon4003 May 31, 2023
Author

thank you so much!! I know it's not OME issue.
But I want any suggestion about receive data after that.
Have a nice day!

getroot May 31, 2023
Maintainer Sponsor

This is an issue where WebRTC communication fails after signaling is complete. Specifically, your WebRTC library is not sending or receiving Stun Binding Requests to OME. It's most likely an incorrect exchange of candidates.

Please refer to OvenRtcTester developed in GO language. This is very similar to yours.

https://github.com/AirenSoft/OvenMediaEngine/blob/master/misc/oven_rtc_tester/OvenRtcTester.go

Answer selected by getroot

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ice_port: Client has expired #1259

toon4003 May 31, 2023

Replies: 1 comment · 6 replies

getroot May 31, 2023 Maintainer Sponsor

toon4003 May 31, 2023 Author

getroot May 31, 2023 Maintainer Sponsor

toon4003 May 31, 2023 Author

getroot May 31, 2023 Maintainer Sponsor

toon4003
May 31, 2023

Replies: 1 comment 6 replies

getroot
May 31, 2023
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toon4003 May 31, 2023
Author

getroot May 31, 2023
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toon4003 May 31, 2023
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