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How too use ovenplayer demo site?

See original GitHub issue

I setup ovenmedia engine through docker on my unraid server I started my steam with OBS. It is working and Ovenmedia engine logs show the connection

I goto demo.ovenplayer.com and enter ws://MYIP:3333/app/stream?transport=tcp and select add source then hit play. It spins for a bit then stops and flashes "connection with low latency terminated unexpectedly ". It then spins again and stops again and the cycle repeats. This is the log from ovenmediaengine server and what it shows.

`Video Track #0: Bypass(true) Bitrate(10.00Mb) codec(1, H264) resolution(3440x1440) framerate(60.00fps) timebase(1/1000) Audio Track #1: Bypass(true) Bitrate(160.00Kb) codec(6, AAC) samplerate(1.0K) format(s16, 16) channel(stereo, 2) timebase(1/1000) Audio Track #2: Bypass(false) Bitrate(128.00Kb) codec(8, OPUS) samplerate(48.0K) format(s16, 16) channel(stereo, 2) timebase(1/48000)[0m [37m[2022-01-22 16:20:47.339] I [SPRTMP-T1935:32] MediaRouter | mediarouter_stream.cpp:54 | Trying to create media route stream: name(stream) id(1037098710)[0m [37m[2022-01-22 16:20:47.339] I [SPRTMP-T1935:32] Monitor | application_metrics.cpp:57 | Create StreamMetrics(stream/e377564b-928b-4213-b98c-b5db440ecdc7/default/#default#app/stream/o) for monitoring[0m [37m[2022-01-22 16:20:47.339] I [SPRTMP-T1935:32] Transcoder | transcoder_stream.cpp:100 | [#default#app/stream(21)] Transcoder input stream has been started. Status : (1) Decoders, (0) Encoders[0m [37m[2022-01-22 16:20:47.362] I [Decaac:42] Transcoder | decoder_aac.cpp:223 | [#default#app/stream(21)] input stream information: [audio] aac (LC), 48000 Hz, stereo, fltp, 117 kbps, timebase: 1/1000, frame_size: 1024[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] WebRTC Publisher | rtc_stream.cpp:279 | Unsupported codec(Audio/AAC) is being input from media track[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] WebRTC Publisher | rtc_stream.cpp:353 | WebRTC Stream has been created : stream/1037098710 Rtx(FALSE) Ulpfec(FALSE) JitterBuffer(FALSE) PlayoutDelay(FALSE min:0 max: 0)[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] Publisher | stream.cpp:181 | WebRTC Publisher Application application has started [stream(1037098710)] stream (MSID : 0)[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] HLS | hls_packetizer.cpp:152 | [#default#app/stream] HLS: Packetizer will be reset: 0[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] Publisher | stream.cpp:181 | HLS Publisher Application application has started [stream(1037098710)] stream (MSID : 0)[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] Publisher | stream.cpp:181 | DASH Publisher Application application has started [stream(1037098710)] stream (MSID : 0)[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] Publisher | stream.cpp:181 | LLDASH Publisher Application application has started [stream(1037098710)] stream (MSID : 0)[0m [37m[2022-01-22 16:20:47.380] I [OutboundWorker:34] Publisher | stream.cpp:181 | OVTPublisher Application application has started [stream(1037098710)] stream (MSID : 0)[0m [37m[2022-01-22 16:20:47.381] I [AW-DASH0:37] DASH | dash_packetizer.cpp:271 | [#default#app/stream] DASH: init_audio.m4s has created[0m [37m[2022-01-22 16:20:51.556] I [AW-DASH0:37] DASH | dash_packetizer.cpp:237 | [#default#app/stream] DASH: init_video.m4s has created[0m [37m[2022-01-22 16:20:59.883] I [AW-LLDASH0:38] LLDASH | cmaf_packetizer.cpp:765 | [0x14fe3800c070] LLDASH segment is ready to stream [#default#app/stream], segment duration: 5.000000s, count: 1[0m [91m[2022-01-22 16:20:59.883] E [AW-LLDASH0:38] LLDASH | cmaf_packetizer.cpp:866 | [#default#app/stream] Because the jitter is too high, playback may not be possible (2022-01-23T00:20:47.380+00:00 => 2022-01-23T00:20:48.380+00:00, 1000ms)

Elapsed: 12503ms (current: 1642897259883ms, start: 1642897247380ms)
Jitter: 4186ms (4186ms - 0ms), correction: 0ms => 1000ms (-1000ms)
Stream delta: 8317ms
Video: last PTS: 12483ms, start: 4166ms, delta: 8317ms
Audio: last PTS: 12480ms, start: 0ms, delta: 12480ms
A-V Sync: 4163 (A: 12480, V: 8317)[0m [91m[2022-01-22 16:21:07.412] E [AW-LLDASH0:38] LLDASH | cmaf_packetizer.cpp:866 | [#default#app/stream] Because the jitter is too high, playback may not be possible (2022-01-23T00:20:48.380+00:00 => 2022-01-23T00:20:49.380+00:00, 2000ms)
Elapsed: 20032ms (current: 1642897267412ms, start: 1642897247380ms)
Jitter: 3182ms (4182ms - 1000ms), correction: 1000ms => 2000ms (-1000ms)
Stream delta: 15850ms
Video: last PTS: 20016ms, start: 4166ms, delta: 15850ms
Audio: last PTS: 19989ms, start: 0ms, delta: 19989ms
A-V Sync: 4139 (A: 19989, V: 15850)[0m [37m[2022-01-22 16:21:08.227] I [AW-DASH0:37] DASH | dash_packetizer.cpp:801 | [#default#app/stream] DASH: Segments are ready to stream, segment duration: 5.000000s, count: 3[0m [37m[2022-01-22 16:21:16.549] I [AW-HLS0:36] HLS | hls_packetizer.cpp:496 | [#default#app/stream] HLS: Segments are ready, segment duration: 5.000000s, count: 3[0m [91m[2022-01-22 16:21:16.549] E [AW-LLDASH0:38] LLDASH | cmaf_packetizer.cpp:866 | [#default#app/stream] Because the jitter is too high, playback may not be possible (2022-01-23T00:20:49.380+00:00 => 2022-01-23T00:20:50.380+00:00, 3000ms)
Elapsed: 29169ms (current: 1642897276549ms, start: 1642897247380ms)
Jitter: 2185ms (4185ms - 2000ms), correction: 2000ms => 3000ms (-1000ms)
Stream delta: 24984ms
Video: last PTS: 29150ms, start: 4166ms, delta: 24984ms
Audio: last PTS: 29141ms, start: 0ms, delta: 29141ms
A-V Sync: 4157 (A: 29141, V: 24984)[0m [37m[2022-01-22 16:21:52.925] I [SPRtcSig-T3333:11] Signalling | rtc_signalling_server.cpp:201 | New client is connected: <ClientSocket: 0x14fe70001130, #22, Connected, TCP, Nonblocking, 192.168.1.1:39723>[0m [37m[2022-01-22 16:21:52.935] I [SPRtcSig-T3333:11] 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)[0m [37m[2022-01-22 16:22:08.271] I [SPRtcSig-T3333:11] WebRTC Publisher | webrtc_publisher.cpp:605 | Stop command received : #default#app/stream/100[0m [37m[2022-01-22 16:22:08.271] I [SPRtcSig-T3333:11] 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)][0m [37m[2022-01-22 16:22:08.271] I [SPRtcSig-T3333:11] Signalling | rtc_signalling_server.cpp:335 | Client is disconnected: <WebSocketClient: 0x14fe70005550, <ClientSocket: 0x14fe70001130, #22, Closed, TCP, Nonblocking, 192.168.1.1:39723>> (#default#app / stream, ufrag: local: CPLhw8, remote: e21f522d)[0m [37m[2022-01-22 16:22:08.271] I [SPRtcSig-T3333:11] WebRTC Publisher | webrtc_publisher.cpp:605 | Stop command received : #default#app/stream/100[0m [37m[2022-01-22 16:22:08.271] I [SPRtcSig-T3333:11] 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)][0m [37m[2022-01-22 16:22:08.271] I [SPRtcSig-T3333:11] Signalling | rtc_signalling_server.cpp:335 | Client is disconnected: <WebSocketClient: 0x14fe70005550, <ClientSocket: 0x14fe70001130, #22, Closed, TCP, Nonblocking, 192.168.1.1:39723>> (#default#app / stream, ufrag: local: CPLhw8, remote: e21f522d)[0m [37m[2022-01-22 16:22:13.895] I [SPRtcSig-T3333:11] Signalling | rtc_signalling_server.cpp:201 | New client is connected: <ClientSocket: 0x14fe70001130, #22, Connected, TCP, Nonblocking, 192.168.1.1:51544>[0m [37m[2022-01-22 16:22:13.903] I [SPRtcSig-T3333:11] 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)[0m [33m[2022-01-22 16:22:44.220] W [DQICETmout:12] ICE | ice_port.cpp:400 | Client (session id: 101) has expired[0m [37m[2022-01-22 16:22:44.220] I [DQICETmout:12] WebRTC Publisher | webrtc_publisher.cpp:671 | IcePort is disconnected. : (#default#app/stream/101) reason(5)[0m [37m[2022-01-22 16:22:44.243] I [SPRtcSig-T3333:11] WebRTC Publisher | webrtc_publisher.cpp:605 | Stop command received : #default#app/stream/101[0m [37m[2022-01-22 16:22:44.243] I [SPRtcSig-T3333:11] 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)][0m [37m[2022-01-22 16:22:44.243] I [SPRtcSig-T3333:11] Signalling | rtc_signalling_server.cpp:335 | Client is disconnected: <WebSocketClient: 0x14fe70004d70, <ClientSocket: 0x14fe70001130, #22, Closed, TCP, Nonblocking, 192.168.1.1:51544>> (#default#app / stream, ufrag: local: 3pReI5, remote: 61c33cb1)[0m [37m[2022-01-22 16:22:46.254] I [SPRtcSig-T3333:11] Signalling | rtc_signalling_server.cpp:201 | New client is connected: <ClientSocket: 0x14fe70001130, #22, Connected, TCP, Nonblocking, 192.168.1.1:20390>[0m [37m[2022-01-22 16:22:46.263] I [SPRtcSig-T3333:11] 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)[0m`

Issue Analytics

State:
Created 2 years ago
Comments:17 (8 by maintainers)

Top GitHub Comments

1reaction

NK0D1NGcommented, Oct 22, 2022

This seems to be the reason why it didn’t work for me in a localhost environment (I used Firefox): https://mediasoup.discourse.group/t/firefox-ice-failed-add-a-stun-server-and-see-about-webrtc-for-more-details/805

0reactions

getrootcommented, Oct 22, 2022

@NK0D1NG Thanks for sharing good information.

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