Error when pushing livestream from OME to another rtmp server ? #720
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When asking a question, please upload the entire log file and the entire configuration file as required by the document template. Otherwise, no one can accurately analyze your problem. |
Here is my log file:
[2022-04-04 04:42:58.541] I [SPRtcSig-T3333:11] MediaRouter | mediarouter_stream.cpp:54 | Trying to create media route stream: name(stream) id(100) [2022-04-04 04:42:58.543] I [SPRtcSig-T3333:11] MediaRouter | mediarouter_stream.cpp:54 | Trying to create media route stream: name(stream) id(569893963) [2022-04-04 04:49:49.219] I [SPRtcSig-T3333:11] Transcoder | transcoder_stream.cpp:1373 | [#default#app/stream(100)] -> [#default#app/stream(569893963)] Transcoder output stream has been deleted. [2022-04-04 04:49:49.220] I [SPICE-T3478:14] ICE | ice_port.cpp:493 | Turn client has disconnected : <ClientSocket: 0x7f3a5c076180, #29, Disconnected, TCP, Nonblocking, 172.17.0.1:53504> |
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The RTMP protocol does not support opus audio codecs. Please try again by adding AAC codec to the encoding option. Thanks. |
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The Send packet error (-32: Broken pipe) log is expected to be disconnected from the RTMP server. Check the cause of the disconnection on the RTMP server. |
[2022-04-04 05:30:17.020] I [SPRtcSig-T3333:11] Transcoder | transcoder_stream.cpp:1373 | [#default#app/stream(101)] -> [#default#app/stream(1663441544)] Transcoder output stream has been deleted. [2022-04-04 05:30:17.021] I [SPRtcSig-T3333:11] Publisher | stream.cpp:259 | [stream(1663441544)] WebRTC Publisher Application stream has been stopped [2022-04-04 05:30:18.155] I [SPRtcSig-T3333:11] MediaRouter | mediarouter_stream.cpp:54 | Trying to create media route stream: name(stream) id(102) [2022-04-04 05:30:18.156] I [SPRtcSig-T3333:11] MediaRouter | mediarouter_stream.cpp:54 | Trying to create media route stream: name(stream) id(3377165238) |
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The above problem is that packets cannot be received from other RTMP servers. Thanks |
My rtmp server is still receiving incoming packets but the stream is always interrupted and not received smoothly. OvenMediaEngine origin * false stun.l.google.com:19302 |
I noticed the incoming rtmp stream is fragmented but not seamless, the stream is split, my rtmp server saves them, they are always 2.3MB in size |
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How is your OME server's CPU usage? |
I play Webrtc with OvenPlayer and it works fine, my CPU goes very high when I push, this must be my problem because I'm just testing on a pretty weak PC, this can be fixed in the file config okay? i just get the webrtc stream and push it to my rtmp server |
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If WebRTC playback works, it's most likely a problem with your RTMP server. Is it correct that OME and RTMP Server are running together on one PC, and the RTMP Server uses a lot of CPU? |
i am testing OME and rtmp server on same PC , my rtmp server converts rtmp to hls adaptive . if not convert the rtmp stream will be fine, but I turned off hls + dash + ... on OME using only WebRTC and RTMP , I don't think it will cost that much when OME just receive WebRTC stream and push to rtmp server , if I push rtmp from OBS my rtmp server works fine |
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You let OME do the encoding. And your RTMP server probably also encodes (Adaptive HLS). Encoding uses a lot of CPU power. Perhaps running encoding on OME and your RTMP server at the same time is over the threshold of your CPU power. Why don't you use HLS in OME? |
thanks for the help , i pushed the rtmp stream to a remote server and it worked fine . Regarding using HLS on OME, since I'm new to OME, I'm not sure if the tasks I need can be done on OME? Can you tell me more about configuring HLS adaptive bitrate playback like using ffmpeg ? I also want to save HLS stream and support DVR |
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A lot of your questions are answered in the documentation and other posts here on GitHub: The Live Source section describes all of the possible input streams (referred to in OME as 'Providers'): https://airensoft.gitbook.io/ovenmediaengine/live-source The Streaming section describes all of the possible playback methods (referred to in OME as 'Publishers'): For stream recording (DVR): https://airensoft.gitbook.io/ovenmediaengine/recording-experiment This roadmap post addresses their upcoming development priorities including ABR for HLS: #548 |
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@bchah I don't believe DVR is the same as recording. At least that's not how I'd use the term. OME supports recording, yes, but not long-playlist-length HLS for being able to seek backward in a live stream. It is one of the key missing features from OME that we have found, since moving from nginx-rtmp I have a related issue with pushing a stream from OME to nginx-rtmp. Randomly the connection seems to drop and restore (very quickly, within a few seconds). There doesn't appear to be anything in the log files for either docker container, but this leads to buffering for any viewers watching via the nginx-rtmp server (which we need for DVR support). Is this a known issue of some sort? |
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This issue has been automatically marked as stale because it has not had recent activity. It will be closed if no further activity occurs. Thank you for your contributions. |
When trying to push the rtmp stream to another server , my rtmp stream always breaks , here is the log i get :
[2022-04-02 04:00:36.454] E [StreamWorker:99] RTMPWriter | rtmp_writer.cpp:393 | Send packet error(-104:Connection reset by peer) [2022-04-02 04:00:36.454] E [StreamWorker:99] RTMPPush | rtmppush_session.cpp:174 | Failed to send packet [2022-04-02 04:00:41.461] W [AW-RTMPPush0:28] RTMPPush | rtmppush_session.cpp:86 | Could not supported codec. track_id:2, codec_id: 8Help me
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