ICE has expired aiortc python client

[XiphosF]

Hello,
I’m developing a Python WebRTC client using the aiortc library. I’m running the default OME configuration locally with Docker, publishing the stream on ws://localhost:3333/app/stream (which works well with a web player).
My implementation follows the OME signaling protocol: it sends the request_offer command, receives the SDP offer with ICE candidates, processes them, and sends back its SDP with ICE candidates.
However, after this exchange, nothing happens except for keep-alive pings, and eventually, the ICE connection expires. Could the issue be related to the SDP or ICE candidates sent by my client?

Here are the OME logs when trying to connect :

[2025-01-13 10:31:44.882] I [SPRtcSig-t3333:10] HTTP.Server | http_server.cpp:149 | Client(<ClientSocket: 0x772240001610, #30, Connected, TCP, Nonblocking, 172.18.0.1:35838>) is connected on *:3333
[2025-01-13 10:31:44.882] I [SPRtcSig-t3333:10] Signalling | rtc_signalling_server.cpp:316 | New client is connected: <ClientSocket: 0x772240001610, #30, Connected, TCP, Nonblocking, 172.18.0.1:35838>
[2025-01-13 10:31:44.927] I [SPRtcSig-t3333:10] Monitor | stream_metrics.cpp:114 | A new session has started playing #default#app/stream on the WebRTC publisher. WebRTC(2)/Stream total(2)/App total(2)
[2025-01-13 10:31:44.927] I [SPRtcSig-t3333:10] ICE | ice_port.cpp:368 | Added session: 86 (ufrag: 6gUsuv:v0gE)
[2025-01-13 10:32:15.458] W [DQICETmout:13] ICE | ice_port.cpp:527 | Agent [Unknow, 86] has expired
[2025-01-13 10:32:15.458] I [DQICETmout:13] ICE | ice_port.cpp:436 | Removed session(86) from ICEPort | ice_seesions_with_id count(1) ice_sessions_with_ufrag(1) ice_sessions_with_address_pair(1)
[2025-01-13 10:32:15.458] I [DQICETmout:13] WebRTC Publisher | webrtc_publisher.cpp:626 | IcePort is disconnected. : (#default#app/stream/5141) reason(6)
[2025-01-13 10:32:15.458] I [DQICETmout:13] WebRTC Publisher | webrtc_publisher.cpp:534 | Stop command received : #default#app/stream/5141
[2025-01-13 10:32:15.458] I [DQICETmout:13] Monitor | stream_metrics.cpp:136 | A session has been stopped playing #default#app/stream on the WebRTC publisher. Concurrent Viewers[WebRTC(1)/Stream total(1)/App total(1)]
[2025-01-13 10:32:15.458] I [DQICETmout:13] Signalling | rtc_signalling_server.cpp:450 | Client is disconnected: HttpConnection(0x772240045610) : WebSocket <ClientSocket: 0x772240001610, #30, Connected, TCP, Nonblocking, 172.18.0.1:35838> TLS(Disabled) (#default#app / stream, ufrag: local: 6gUsuv, remote: v0gE)
[2025-01-13 10:32:15.489] I [SPRtcSig-t3333:10] HTTP.Server | http_server.cpp:216 | Client(<ClientSocket: 0x772240001610, #30, Closed, TCP, Nonblocking, 172.18.0.1:35838>) has been disconnected by *:3333

And the logs of my python client for the ICE part:

DEBUG:aiortc.rtcpeerconnection:RTCPeerConnection() setRemoteDescription(offer)
DEBUG:root:Added initial ICE candidate from server: candidate:8921547360 1 UDP 50 172.18.0.2 10003 typ host
DEBUG:root:Added initial ICE candidate from server: candidate:2951473608 1 UDP 50 91.165.226.167 10003 typ host
DEBUG:aiortc.rtcpeerconnection:RTCPeerConnection() setLocalDescription(answer)
DEBUG:aiortc.rtcpeerconnection:RTCPeerConnection() iceGatheringState new -> gathering
DEBUG:aioice.ice:Connection(0) protocol(0) connection_made(<_SelectorDatagramTransport fd=7 read=idle write=<idle, bufsize=0>>)
DEBUG:aioice.ice:Connection(0) protocol(1) connection_made(<_SelectorDatagramTransport fd=8 read=idle write=<idle, bufsize=0>>)
DEBUG:aioice.ice:Connection(0) protocol(2) connection_made(<_SelectorDatagramTransport fd=9 read=idle write=<idle, bufsize=0>>)
DEBUG:aioice.ice:Connection(0) protocol(1) > ('74.125.250.129', 19302) Message(message_method=Method.BINDING, message_class=Class.REQUEST, transaction_id=b'\xad\xd2\r\xff\xd3\x8a`b\xbc=\xe0o')
DEBUG:aioice.ice:Connection(0) protocol(0) > ('74.125.250.129', 19302) Message(message_method=Method.BINDING, message_class=Class.REQUEST, transaction_id=b'\xe1=\xde\x0e\xaf\xbdK\xe8\x0e\xf4oQ')
DEBUG:aioice.ice:Connection(0) protocol(2) > ('74.125.250.129', 19302) Message(message_method=Method.BINDING, message_class=Class.REQUEST, transaction_id=b'\x05\tv\xe1\xa0\xfa\xd6\x10\x952H\xce')
DEBUG:aioice.ice:Connection(0) protocol(0) < ('74.125.250.129', 19302) Message(message_method=Method.BINDING, message_class=Class.RESPONSE, transaction_id=b'\xe1=\xde\x0e\xaf\xbdK\xe8\x0e\xf4oQ')
DEBUG:aioice.ice:Connection(0) protocol(1) < ('74.125.250.129', 19302) Message(message_method=Method.BINDING, message_class=Class.RESPONSE, transaction_id=b'\xad\xd2\r\xff\xd3\x8a`b\xbc=\xe0o')
DEBUG:aioice.ice:Connection(0) protocol(2) < ('74.125.250.129', 19302) Message(message_method=Method.BINDING, message_class=Class.RESPONSE, transaction_id=b'\x05\tv\xe1\xa0\xfa\xd6\x10\x952H\xce')
DEBUG:aiortc.rtcpeerconnection:RTCPeerConnection() iceGatheringState gathering -> complete
DEBUG:websockets.client:> TEXT '{"command": "answer", "id": 1331170613, "sdp": ..., "sdpMLineIndex": 0}]}' [2421 bytes]
INFO:root:Sent answer to server
DEBUG:aiortc.rtcicetransport:RTCIceTransport(controlled) - new -> checking
DEBUG:aiortc.rtcpeerconnection:RTCPeerConnection() iceConnectionState new -> checking
DEBUG:aiortc.rtcpeerconnection:RTCPeerConnection() connectionState new -> connecting
INFO:root:ICE connection state changed to checking
DEBUG:websockets.client:< TEXT '{"command":"notification","message":{"auto":fal...}}]},"type":"playlist"}' [240 bytes]
DEBUG:websockets.client:< TEXT '{"command":"notification","message":{"auto":fal...e":"rendition_changed"}' [105 bytes]
DEBUG:root:Received message from server: {'command': 'notification', 'message': {'auto': False, 'file_name': 'webrtc_default', 'name': 'webrtc_default', 'renditions': [{'name': 'default', 'video_track': {'id': 0, 'name': 'bypass_video', 'type': 'Video', 'video': {'bypass': True}}}]}, 'type': 'playlist'}
DEBUG:root:Received message from server: {'command': 'notification', 'message': {'auto': False, 'rendition_name': 'default'}, 'type': 'rendition_changed'}
DEBUG:websockets.client:% sending keepalive ping
DEBUG:websockets.client:> PING 7c 2c dc c0 [binary, 4 bytes]
DEBUG:websockets.client:< PONG 7c 2c dc c0 [binary, 4 bytes]
DEBUG:websockets.client:% received keepalive pong
DEBUG:websockets.client:< PING 'OvenMediaEngine' [text, 15 bytes]
DEBUG:websockets.client:> PONG 'OvenMediaEngine' [text, 15 bytes]
DEBUG:websockets.client:< EOF
DEBUG:websockets.client:< EOF
DEBUG:websockets.client:> EOF
DEBUG:websockets.client:= connection is CLOSED
DEBUG:websockets.client:x half-closing TCP connection
ERROR:root:Error during execution: no close frame received or sent

The sdp + ICE answer of my client looks like this :

{"command": "answer", "id": 1176309753, "sdp": {"type": "answer", "sdp": "v=0\r\no=- 3945753770 3945753770 IN IP4 0.0.0.0\r\ns=-\r\nt=0 0\r\na=group:BUNDLE HfRo5j\r\na=msid-semantic:WMS *\r\nm=video 51641 UDP/TLS/RTP/SAVPF 98\r\nc=IN IP4 172.30.2.190\r\na=recvonly\r\na=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=mid:HfRo5j\r\na=msid:45993fca-460e-4fea-ac9f-0d706affb40a 4888f728-746a-4a2f-b8b3-5f2f55e38882\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=rtcp-mux\r\na=ssrc:3735313808 cname:3d218145-25b2-4bf6-88e8-fe5a3aa0ba10\r\na=rtpmap:98 H264/90000\r\na=rtcp-fb:98 goog-remb\r\na=fmtp:98 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=candidate:6c81096a932652ca9dc8eb76428007b8 1 udp 2130706431 172.30.2.190 51641 typ host\r\na=candidate:95a5144d22d827ac85a6c13daea75649 1 udp 2130706431 172.18.0.1 54257 typ host\r\na=candidate:580b5f32035da7f14a30ea7a8d826c67 1 udp 2130706431 172.17.0.1 60400 typ host\r\na=candidate:0ad8db753210ce9554cf251a9e51e792 1 udp 1694498815 194.254.113.72 60400 typ srflx raddr 172.17.0.1 rport 60400\r\na=candidate:a9f9bab060c86712caf06907a08d3373 1 udp 1694498815 194.254.113.72 51641 typ srflx raddr 172.30.2.190 rport 51641\r\na=candidate:297898e4296c74660bb3ab65d8da90ec 1 udp 1694498815 194.254.113.72 54257 typ srflx raddr 172.18.0.1 rport 54257\r\na=end-of-candidates\r\na=ice-ufrag:QtVw\r\na=ice-pwd:Y0YilxiP6gAG7oI0UaCtsb\r\na=fingerprint:sha-256 E8:63:4C:D6:E1:58:32:68:6B:BD:E2:16:7F:13:15:1F:7E:B0:07:2E:3B:D5:62:CF:99:C8:16:80:B6:4C:3E:F2\r\na=setup:active\r\n"}, "candidates": [{"candidate": "candidate:6c81096a932652ca9dc8eb76428007b8 1 udp 2130706431 172.30.2.190 51641 typ host", "sdpMLineIndex": 0}, {"candidate": "candidate:95a5144d22d827ac85a6c13daea75649 1 udp 2130706431 172.18.0.1 54257 typ host", "sdpMLineIndex": 0}, {"candidate": "candidate:580b5f32035da7f14a30ea7a8d826c67 1 udp 2130706431 172.17.0.1 60400 typ host", "sdpMLineIndex": 0}, {"candidate": "candidate:0ad8db753210ce9554cf251a9e51e792 1 udp 1694498815 194.254.113.72 60400 typ srflx raddr 172.17.0.1 rport 60400", "sdpMLineIndex": 0}, {"candidate": "candidate:a9f9bab060c86712caf06907a08d3373 1 udp 1694498815 194.254.113.72 51641 typ srflx raddr 172.30.2.190 rport 51641", "sdpMLineIndex": 0}, {"candidate": "candidate:297898e4296c74660bb3ab65d8da90ec 1 udp 1694498815 194.254.113.72 54257 typ srflx raddr 172.18.0.1 rport 54257", "sdpMLineIndex": 0}]}

[getroot]

I am having trouble getting a Stun Request from a client in OME.

Is OME's IceCandidates set to an IP that can communicate with the client?

<WebRTC>
				<Signalling>
					<Port>${env:OME_WEBRTC_SIGNALLING_PORT:3333}</Port>
					<TLSPort>${env:OME_WEBRTC_SIGNALLING_TLS_PORT:3334}</TLSPort>
					<WorkerCount>1</WorkerCount>
				</Signalling>
				<IceCandidates>
					<IceCandidate>${env:OME_HOST_IP:*}:${env:OME_WEBRTC_CANDIDATE_PORT:10000-10004/udp}</IceCandidate>
					<TcpRelay>${env:OME_HOST_IP:*}:${env:OME_WEBRTC_TCP_RELAY_PORT:3478}</TcpRelay>
					<TcpForce>true</TcpForce>
					<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
				</IceCandidates>
			</WebRTC>

[XiphosF]

I believe all the ports on the OME server are configured correctly, as the HTML player receives the stream without issues and i configure the turn server in my client from the one send by OME. However, after some research, I suspect the problem might lie in the fact that aiortc does not support TCP candidates (see aiortc/aiortc#443), while OME provides TCP ICE candidates.

[getroot]

OME does not provide TCP candidate. It just has a built-in turn/tcp server, so it connects end-to-end tcp.

If you have set the turn server address provided by OME to 'iceServers' (I don't know which interface is for this in aiortc), you should check the following:

The IP:Port you set for TcpRelay is the built-in turn server. Can your python client access the address below? The reason I ask this again is that your HTML player and your python client may not be running on the same host.

${env:OME_HOST_IP:*}:${env:OME_WEBRTC_TCP_RELAY_PORT:3478}

OME logs show that STUN Request is not received, and aiortc also seems to be failing the STUN connection.

[getroot]

Our performance tester client is based on Pion. It may be helpful to you.
https://github.com/AirenSoft/OvenMediaEngine/tree/master/misc/oven_rtc_tester

[XiphosF]

Hi, thanks for all your answers, I've finally switched to Gstreamer python binding instead of aiortc and it worked well !