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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...

Unfortunately, VOD does not support pausing at all. You can only start VOD live stream from the beginning of the file or stop it. A workaround is to split VOD file to a small fragments and play them subsequentally, in this case you can stop playback and start it again from a closest fragment...

Good day. The sample file contains only webm header and no media data (has 0:00:00 duration). This can occur if there was no media data in the stream published, or no key frames. The recording can start only from key frame, so seems like ther were no keyframes in the stream during this time.

Good day. You can use REST API: VOD capture management with REST API /vod/startup to start capture from VOD /mixer/add to add VOD stream to mixer /vod/terminate to stop VOD stream

Добрый день. В дальнейшем, пожалуйста, направляйте отчеты или ссылки на них только с помощью формы. Сообщение пришлось удалить, т.к. отчет, доступный по прямой ссылке, может содержать данные лицензии. В логах видно, что перед перезапуском остановилась запись только одного потока, причем только...

Now, most of streaming services require to send metadata. But, if there is no metadata sent, some media servers (Wowza, WCS for example) can get picture parameters directly from media packets. However, seems like Youtube does not support this. So you have to disable RTMP buffering to workaround...

We made some tests almost an year ago. Up to 10 streams 320x240 may be played on the same page on Intel Core i5 8gen (8 HT cores) with 20 Gb RAM. But you have to play WebRTC: MSE can be played only 1 stream per session, so it is required to set up SSL certificates and open the page via HTTPS. If...

We tested republishing from your server. There are two issues in server configuration: 1. The following parameter transcoding_disabled=true prevents any transcoding. But Opus audio must be transcoded to AAC for republishing, so audio is not sent to RTMP server. Please enable transcoding...

Good day. We successfully tested WebRTC as RTMP republishing on your server. It's working, stream is publishing and republishing to our demo server: Please make sure you're opening the page using HTTPS (this is the only case Webrtc will work). Also please make sure you're using a correct stream...

Good day. The first stream we see in logs provided seems to be captured successfully But you're opening the page via HTTP. not HTTP, so WebRTC does not work Please use HTTPS to open a page in browser. Also, we can't reproduce the problem in Player example...

Please also clarify: do you play 15 streams simyltaneously: 1. from one page in browser? 2. from one workstation (PC/Mac) in 15 tabs of one browser? 3. from 15 workstation (one strem per one PC)? For both cases 1 and 2, seems local CPU resources are not enough to play, so local browser...

Sorry, it's a mistype. Should be sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver check_update and sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver update respectively. Please see the doc here.

Hello How exactly do you play 15 streams. We need a code sample, screenshots, full browser console logs in text form. 5 lines of js logs are definitely not enough.

Good day. Please reproduce the issue in Two Way Streaming example https://wcs:8444/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html. If this is not reproduced, please modify the example code minimally to reproduce the issue and send the code using this form.