Search results

This is major addition. Because if you set width and height you force transcoding to the new resolution. We will check how it works in our env.

Please provide full flashphoner.log file via this form Make sure the logfile contains REST requests. If it does not contain REST requests, please gather log of application which send such requests.

Hello Are you trying to change stream resolution from 1280 x 720 to 720 x 1280 ? Resolution input width: 1280 height: 720 Resolution output width: 720 height: 1280

Hello Please check out this doc https://docs.flashphoner.com/display/WCS52EN/Bitrate+management+when+capturing+WebRTC+stream+in+browser Server Global server-side settings should work either for Android, iOS, and Web SDK webrtc_sdp_min_bitrate_bps=3000000 webrtc_sdp_max_bitrate_bps=7000000...

Hello 1. /pull/pull works between WCS and WCS only. If you do /pull/pull and set any other endpoint such as instagram or youtube, this won't be working. 2. If you do /push/startup it should be working either via localhost or via remote server. /push/startup works via RTMP only. So if remote...

Hello We would need a report to proceed with investigation https://docs.flashphoner.com/display/WCS52EN/Preparing+an+error+report#Preparinganerrorreport-Gettinglogswithreport.shscript This report should show the cause of CPU load per CPU thread. Before making report make sure jstack tool is...

Thanks. We were able to locate the gray background on your video. We would need a bit more time to reproduce such effect on our env. 0:04 second 0:22 second 0:25 second

Hello Error resistance during recording has been improved in 5.2.1236 version Releases https://docs.flashphoner.com/display/WCS52EN/Release+notes Recording docs https://docs.flashphoner.com/display/WCS52EN/Stream+recording#Streamrecording-Recordingstoppingonerrors Before this improvement...

WebRTC over UDP is faster than TCP, but UDP is less packet loss proof than TCP. So on good stable internet connections UDP is recommended to use, but on unstable one TCP is preferable. You can switch transport on client side in Android and iOS SDKs. You cannot synchronize two different stream...

Здравствуйте. Похожая проблема была исправлена, начиная со сборки 5.2.1252 https://docs.flashphoner.com/display/WCS52EN/Release+notes Соответственно, нужно обновиться до этой версии или выше.

Hello 1. Try to update. The build 5.2.1054 is October 2021. Looking quite old. Try to update to the latest available release https://docs.flashphoner.com/display/WCS52EN/Release+notes Make sure you have backup before update and you know how to rollback all changes quickly. 2. Switch to TCP...

Yes. /pull/pull https://docs.flashphoner.com/display/WCS52EN/From+another+WCS+server+via+WebRTC /pull/push https://docs.flashphoner.com/display/WCS52EN/To+another+WCS+server+via+WebRTC

We cannot reproduce a problem on both servers: websocket connection is establishing successfully (with Chrome browsers warning because we connected by IP), stream publishing and playback are working: 1st: 2nd: Unfortunately we do not know these servers domain names to check the connection. The...

Good day. Please provide SSH access to the server using this form. Transcoding will work in any WCS build unless it is disabled in flashphoner.properties or server capacity is not enough to encode incoming streams.

Good day. Please look at Conference example (code on GitHub). You can locally mute the participants stream when playing it. Just set stream option unmutePlayOnStart: true (sample code). Or, if you want to mute stream on publisher side, you should use stream.muteAudio() method. In this case, you...

Please clarify your flow: 1. Start the mixer by REST API /mixer/startup 2. Play the mixer stream (should be black box by default) 3. Publish WebRTC stream from camera 4. Add the stream to mixer by REST API /mixer/add Is that right? On what step do you see a gray box? We can't reproduce this in a...

Please check if WebRTC can be published in Two Way Streaming example https://wcs:8888/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html using VPN. If not (returns Failed by ICE timeout, try to switch the client to TCP transport session.createStream({ name: streamName...

Добрый день. Попробуйте на стороне сервера отключить использование дополнительных кодеков allow_outside_codecs=false Также, если Ваш SIP-сервер поддерживает TCP транспорт для SIP сигналинга, включите на стороне сервера sip_force_tcp=true Кроме того, некоторые провайдеры частично блокируют DTLS...

Good day. Seems like the camera sends VSIP in media packets, not H264. Please provide a public RTSP URL using this form. If this is not possible, please collect the report including traffic dump as described here: 1. Enable client debug logs client_log_level=DEBUG and restart WCS sudo systemctl...

Please enable periodic FIR request periodic_fir_request=true periodic_fir_request_interval=2000 In this case, server asks client browser to send key frame every 2 seconds. Also make sure that your server has capacity enough for mixer to work: at least 2 CPU cores per one mixer + 2 cores...