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No, there are no any restrictions at server side (but your provider may apply them). Please check if publishing/playback is working in browser on the same device using Two Way Streaming example.

Please check if UDP media ports (31001-32000) are available on server from client: Port routing checking Also you can try to use TCP transport for WebRTC if React Native WebRTC implementation supports it: Publishing and playing stream via WebRTC over TCP Please also check if publishing/playback...

When publishing a stream, publishStream should be sent, then setRemoteSDP and notifyStreamStatusEvent are received. Seems like you're receiving the messages above, so signaling is working. But, if onconnectionstatechange returns failed state, it means WebRTC connection is not established. This...

Actually Web SDK is full implementation of Raw Websocket API + WebRTC https://github.com/flashphoner/flashphoner_client MediaSessionId is UUID generated on the JavaScript end https://github.com/flashphoner/flashphoner_client/blob/wcs_api-2.0/src/flashphoner-core.js#L1632 var id_ = uuid_v1()...

Please clarify how do you publish a stream? What resolution and bitrate do you use? Try to check publish/playback channel quality: Publisher and player channel quality control If the quality is BAD, try to use TCP for publishing/playback, use lower resolution/bitrate to conform the...

Good day. The possible reason is an uneven key frames flow in media stream. If you're publishing RTMP stream from OBS, you should set key frame interval like this: If you're publishing WebRTC from browser, set the following WCS parameters in flashphoner.properties file...

This is a rare case to stop an RTMP publishinig stream without subscribers. So we have no plans to add an option for this. However, there is a such option for pulled RTSP or RTMP streams.

Good day. You can check if there are stream subscribers by REST API /stream/find: POST /rest-api/stream/find HTTP/1.1 Host: wcs:8081 Content-Type: application/json { "name":"test", "published":false } The query returns 404 Not Found if there are no subscribers to the stream. So you...

We raised the ticket WCS-3598 for this feature request. Note that it may take a time, so /stream/find REST API polling seems to be a quick workaround.

Again, the file has a wrong structure: free, then mdat, finally moov but moov must be the first for progressive downloading and playback If you need to play WCS recordings from S3 bucket, you should set the moov atom to be placed first: moov atom placement configuration in recording metadata

Good day. You can check if a stream is recording by /stream/find REST API query. It will return record: true for stream which is recording now.

Please try to publish the stream from OBS with settings above to WCS directly How it is playing as WebRTC in Media Devices example (Chrome browser)? If it is playing good, add it to mixer. Does it still playing good? If both anwsers are yes, try to publish the stream with the same settings to...

Seems like network issue between the server and S3 bucket endpoint which leading to bitstream break. Try to use lower resolution/bitrate, use the server and the bucket in the same region. If this does not help, please provide the file (or a download link to it if its size more than 30 M) using...

OBS should be enough. Max B-frames options should be set to 0. NVIDIA X264

We checked the stream. It has B-frames and AV synchronization problems: Even in Player example it's jerking. Also, bitrate and FPS are jumping: WCS does not support B-frames because Chrome still cannot play them smoothly. So the problem is in stream itself, not in mixer. You should reencode...

We fixed a stream termination by name (and other parameters, see Stream filtering by parameters) in WCS build 5.2.1386.

Good day. We can't reprtoduce the issue in WCS latest build 5.2.1386: a pulled RTMP stream is playing with the same quality in mixer as directly played. So try to update WCS, then please check if the pulled RTMP stream is playing smoothly directly from WCS. Please test in Media Devices example...

We fixed the issue with some RTMP streams pulling (ticket WCS-3587) in build 5.2.1383 (also available on demo server).

Futher, please create a your own topic rather than bump up an old one.

Разница в том, что при стриминге с OBS битрейт будет стабильным, а при републикации WebRTC стрима как RTMP битрейт будет зависеть от битрейта источника. По приведенным Вами ранее метрикам, у Вас плохой канал от ПК до WCS, которого не хватает на заданный битрейт. Либо, другой вариант. Вы...