Connection Failed when upgrading to newest version (2.5.2 - 2.5.7)

[padi-dev-anhntn]

Hi,
I use 2.5.1 and it works well, I start streaming from mobile and can see the video streaming on the website.
I want to upgrade to newer version(2.5.2 - 2.5.7) to remove the noise when streaming using rtmpStream.setTimestampMode(TimestampMode.BUFFER, TimestampMode.BUFFER);
But when I upgrade to newer version, it had errors like this, and I can't see the video streaming on the website.
'rtmps://abc-live-20-01-abcd.rtmp.livekit.cloud/x/abcdef'

/Handshake( 6589): writing C0
I/Handshake( 6589): C0 write successful
I/Handshake( 6589): writing C1
I/Handshake( 6589): writing time 1740476260 to c1
I/Handshake( 6589): writing zero to c1
I/Handshake( 6589): writing random to c1
I/Handshake( 6589): C1 write successful
I/Handshake( 6589): reading S0
I/Handshake( 6589): read S0 successful
I/Handshake( 6589): reading S1
E/RtmpClient( 6589): connection error
E/RtmpClient( 6589): kotlinx.coroutines.channels.ClosedReceiveChannelException: Unexpected EOF: expected 1434 more bytes
E/RtmpClient( 6589): 	at io.ktor.utils.io.ByteBufferChannel.readFullySuspend(ByteBufferChannel.kt:623)
E/RtmpClient( 6589): 	at io.ktor.utils.io.ByteBufferChannel.access$readFullySuspend(ByteBufferChannel.kt:23)
E/RtmpClient( 6589): 	at io.ktor.utils.io.ByteBufferChannel$readFullySuspend$3.invokeSuspend(Unknown Source:16)
E/RtmpClient( 6589): 	at kotlin.coroutines.jvm.internal.BaseContinuationImpl.resumeWith(ContinuationImpl.kt:33)
E/RtmpClient( 6589): 	at kotlinx.coroutines.DispatchedTask.run(DispatchedTask.kt:100)
E/RtmpClient( 6589): 	at kotlinx.coroutines.internal.LimitedDispatcher$Worker.run(LimitedDispatcher.kt:113)
E/RtmpClient( 6589): 	at kotlinx.coroutines.scheduling.TaskImpl.run(Tasks.kt:89)
E/RtmpClient( 6589): 	at kotlinx.coroutines.scheduling.CoroutineScheduler.runSafely(CoroutineScheduler.kt:586)
E/RtmpClient( 6589): 	at kotlinx.coroutines.scheduling.CoroutineScheduler$Worker.executeTask(CoroutineScheduler.kt:820)
E/RtmpClient( 6589): 	at kotlinx.coroutines.scheduling.CoroutineScheduler$Worker.runWorker(CoroutineScheduler.kt:717)
E/RtmpClient( 6589): 	at kotlinx.coroutines.scheduling.CoroutineScheduler$Worker.run(CoroutineScheduler.kt:704)
E/duc2111 ( 6589): onConnectionFailed = Error configure stream, Unexpected EOF: expected 1434 more bytes
W/VideoCapabilities( 6589): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 6589): Unrecognized profile 2130706434 for video/avc
D/FlutterWebRTCPlugin( 6589): onIceGatheringChangeGATHERING
D/FlutterWebRTCPlugin( 6589): onIceCandidate
I/chatty  ( 6589): uid=10950(app.boosty.live.boosty_live_app.stg) signaling_threa identical 4 lines
2
D/FlutterWebRTCPlugin( 6589): onIceCandidate
I/chatty  ( 6589): uid=10950(app.boosty.live.boosty_live_app.stg) signaling_threa identical 1 line
2
D/FlutterWebRTCPlugin( 6589): onIceCandidate
I/chatty  ( 6589): uid=10950(app.boosty.live.boosty_live_app.stg) signaling_threa identical 1 line
5
D/FlutterWebRTCPlugin( 6589): onIceCandidate
2
I/flutter ( 6589): isCameraOn camera - true
W/FlutterWebRTCPlugin( 6589): audioFocusChangeListener [Earpiece(name=Earpiece)] Earpiece(name=Earpiece)
W/FlutterWebRTCPlugin( 6589): audioFocusChangeListener [Speakerphone(name=Speakerphone), Earpiece(name=Earpiece)] Speakerphone(name=Speakerphone)
D/MediaConstraintsUtils( 6589): mandatory constraints are not a map
D/MediaConstraintsUtils( 6589): optional constraints are not an array
D/FlutterWebRTCPlugin( 6589): onConnectionChangeCONNECTING
D/FlutterWebRTCPlugin( 6589): onSelectedCandidatePairChanged
D/FlutterWebRTCPlugin( 6589): onIceGatheringChangeCOMPLETE
D/FlutterWebRTCPlugin( 6589): onConnectionChangeCONNECTED
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: initRecording(sampleRate=48000, channels=1)
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: byteBuffer.capacity: 960
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: AudioRecord.getMinBufferSize: 3840
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: bufferSizeInBytes: 7680
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: createAudioRecordOnMOrHigher
W/AudioRecord( 6589): dead IAudioRecord, creating a new one from obtainBuffer()
I/org.webrtc.Logging( 6589): WebRtcAudioEffectsExternal: enable(audioSession=52489)
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: AudioRecord: session ID: 52489, channels: 1, sample rate: 48000
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: AudioRecord: buffer size in frames: 3840
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: Number of active recording sessions: 0
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: startRecording
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: scheduleLogRecordingConfigurationsTask
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: AudioRecordThread@[name=AudioRecordJavaThread, id=18174]
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: doAudioRecordStateCallback: START
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: Number of active recording sessions: 2
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: AudioRecordingConfigurations: 
E/AudioRecordingConfiguration( 6589): Couldn't find device for recording, did recording end already?
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal:   client audio source=MIC, client session id=52481 (52489)
I/org.webrtc.Logging( 6589):   Device AudioFormat: channel count=1, channel index mask=0, channel mask=IN_MONO, encoding=PCM_16BIT, sample rate=48000
I/org.webrtc.Logging( 6589):   Client AudioFormat: channel count=2, channel index mask=0, channel mask=IN_STEREO, encoding=PCM_16BIT, sample rate=32000
E/AudioRecordingConfiguration( 6589): Couldn't find device for recording, did recording end already?
I/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal:   client audio source=VOICE_COMMUNICATION, client session id=52489 (52489)
I/org.webrtc.Logging( 6589):   Device AudioFormat: channel count=1, channel index mask=0, channel mask=IN_MONO, encoding=PCM_16BIT, sample rate=48000
I/org.webrtc.Logging( 6589):   Client AudioFormat: channel count=1, channel index mask=0, channel mask=IN_MONO, encoding=PCM_16BIT, sample rate=48000
2
E/AudioRecordingConfiguration( 6589): Couldn't find device for recording, did recording end already?
E/org.webrtc.Logging( 6589): WebRtcAudioRecordExternal: verifyAudioConfig: FAILED

My initialization:

rtmpStream = new RtmpStream(activity, this, videoSource, new MicrophoneSource());
rtmpStream.getGlInterface().setAutoHandleOrientation(true);
rtmpStream.getStreamClient().setBitrateExponentialFactor(0.5f);
rtmpStream.setTimestampMode(TimestampMode.BUFFER, TimestampMode.BUFFER);

rtmpStream.prepareAudio(32000, true, 128 * 1000);
rtmpStream.prepareVideo(1280, 720, 1200* 1000);

 rtmpStream.startStream('rtmps://abc-live-20-01-abcd.rtmp.livekit.cloud/x/abcdef');

 rtmpStream.getGlInterface().setIsPortrait(true);
 rtmpStream.getGlInterface().setCameraOrientation(0);
 rtmpStream.startPreview(surface, resolutionPreset.getHeight(), resolutionPreset.getWidth());

Thanks.

[pedroSG94]

Hello,

I'm testing using that url with version 2.5.7 and version 2.5.1 and I receive the same error. Can you share a valid url to test?
Maybe it is related with the server so I need the same server to reproduce it or a way to configure the server like you to reproduce the error.

[padi-dev-anhntn]

Hi @pedroSG94 ,
Wait for me to configure the valid url to you.
But is it any way related to Url Parser?
Here's format of the url
rtmps://abc-live-dev-aBCDef1gH.rtmp.livekit.cloud/x/aBcd1EfG2H

And it didn't work from this commit.
be98bef
Thanks.

[padi-dev-anhntn]

Hi @pedroSG94 ,

Update:
I use rtmp:// instead of rtmps:// and it works on 2.5.2 - 2.5.7
But I still want to use rtmps://

Thanks.

[pedroSG94]

Hello,

It is maybe related with ktor. Can you test with master branch which contain an update of ktor? compile the project and use the last commit.

[pedroSG94]

But is it any way related to Url Parser?

No, if it is working with rtmp the parser is working correctly

[padi-dev-anhntn]

Hi @pedroSG94 ,
I tried last commit acdb4be

it shows connection error java.io.EOFException: Channel is already closed

I/Handshake( 4851): writing C0
I/Handshake( 4851): C0 write successful
I/Handshake( 4851): writing C1
I/Handshake( 4851): writing time 1740736706 to c1
I/Handshake( 4851): writing zero to c1
I/Handshake( 4851): writing random to c1
I/Handshake( 4851): C1 write successful
I/Handshake( 4851): reading S0
W/VideoCapabilities( 4851): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706434 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706433 for video/avc
W/VideoCapabilities( 4851): Unrecognized profile 2130706434 for video/avc
I/Handshake( 4851): read S0 successful
I/Handshake( 4851): reading S1
D/FlutterWebRTCPlugin( 4851): onIceGatheringChangeGATHERING
D/FlutterWebRTCPlugin( 4851): onIceCandidate
I/chatty  ( 4851): uid=10952(app.boosty.live.boosty_live_app.stg) signaling_threa identical 3 lines
2
D/FlutterWebRTCPlugin( 4851): onIceCandidate
E/RtmpClient( 4851): connection error
E/RtmpClient( 4851): java.io.EOFException: Channel is already closed
E/RtmpClient( 4851): 	at io.ktor.utils.io.ByteReadChannelOperationsKt.readFully(ByteReadChannelOperations.kt:523)
E/RtmpClient( 4851): 	at io.ktor.utils.io.ByteReadChannelOperationsKt$readFully$1.invokeSuspend(Unknown Source:14)
E/RtmpClient( 4851): 	at kotlin.coroutines.jvm.internal.BaseContinuationImpl.resumeWith(ContinuationImpl.kt:33)
E/RtmpClient( 4851): 	at kotlinx.coroutines.DispatchedTask.run(DispatchedTask.kt:100)
E/RtmpClient( 4851): 	at kotlinx.coroutines.internal.LimitedDispatcher$Worker.run(LimitedDispatcher.kt:113)
E/RtmpClient( 4851): 	at kotlinx.coroutines.scheduling.TaskImpl.run(Tasks.kt:89)
E/RtmpClient( 4851): 	at kotlinx.coroutines.scheduling.CoroutineScheduler.runSafely(CoroutineScheduler.kt:586)
E/RtmpClient( 4851): 	at kotlinx.coroutines.scheduling.CoroutineScheduler$Worker.executeTask(CoroutineScheduler.kt:820)
E/RtmpClient( 4851): 	at kotlinx.coroutines.scheduling.CoroutineScheduler$Worker.runWorker(CoroutineScheduler.kt:717)
E/RtmpClient( 4851): 	at kotlinx.coroutines.scheduling.CoroutineScheduler$Worker.run(CoroutineScheduler.kt:704)
E/onConnectionFailed ( 4851): onConnectionFailed = Error configure stream, Channel is already closed
3
D/FlutterWebRTCPlugin( 4851): onIceCandidate
I/chatty  ( 4851): uid=10952(app.boosty.live.boosty_live_app.stg) signaling_threa identical 1 line
4
D/FlutterWebRTCPlugin( 4851): onIceCandidate
I/flutter ( 4851): isCameraOn camera - true
2
D/FlutterWebRTCPlugin( 4851): onIceCandidate
W/FlutterWebRTCPlugin( 4851): audioFocusChangeListener [Earpiece(name=Earpiece)] Earpiece(name=Earpiece)
W/FlutterWebRTCPlugin( 4851): audioFocusChangeListener [Speakerphone(name=Speakerphone), Earpiece(name=Earpiece)] Speakerphone(name=Speakerphone)
D/MediaConstraintsUtils( 4851): mandatory constraints are not a map
D/MediaConstraintsUtils( 4851): optional constraints are not an array

[pedroSG94]

Can you share me a valid endpoint to connect? I'm no able to connect using the old version 2.5.1

[pedroSG94]

I was able to create my own livekit server and reproduce the problem. I will work in this and release a fix

[padi-dev-anhntn]

Thank you so much ♥️

[pedroSG94]

Hello,

I'm working in this but this seem a bug in ktor using tls socket. I can fix the problem using java.io blocking api for socket like before but this could have others problems blocking the send sometimes. I'm looking the best way to do it but I'm not sure how to do it for now.