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Seems like the rsync operation consumes most of channel bandwidth, so freezes andframe losses occur in streams published. According to mixer settings, you're using non-realtime mixer mixer_realtime=false ... mixer_out_buffer_enabled=true mixer_out_buffer_initial_size=3500...

Usually, mobile applications uses WebView to play HLS. So it depends on WebView implementation on a certain platform, but it behaviour should not differ from default browser. If, however, you 've created mobile application from scratch including HTTP custom implementation, you have to implement...

Добрый день. Используйте функцию Stream.unmuteRemoteAudio(), например stream = session.createStream(options).on(STREAM_STATUS.PENDING, function (stream) { ... }).on(STREAM_STATUS.PLAYING, function (stream) { stream.unmuteRemoteAudio(); ... })...

This is Firefox bug, we reported it and they will fix it in 79 or 80 https://bugzilla.mozilla.org/show_bug.cgi?id=1652426#c7 The workaround is to play audio only stream in Firefox with constraints constraints: {audio: true, video: false}

It didn't gather memory dump and jstack (utility link is supposed to be configured). For the playback issue, logs and configs may be sufficient, so please send the generated report.

No, it's our custom implementation

We raised internal ticket WCS-2805 and will let you know about results. We do not provide ETA for new feature requests, because it depends on user votes for the feature. Please read this thread for example. But in iOS case, you should link with libwebrtc, use hints from StackOverflow and...

Unfortunately, no. This is Chrome oversecurity issue. The domain name, protocol and port should be exactly the same as request source.

MCU view with screen sharing is the same for all participants including publisher. Publisher can attach to PC and share a second screen if you do not wish to see its own browser page. Enable the following option mixer_display_stream_name=true and name streams as users, for example user1#room1

Good day. Check if the problem is reproduced on Two Way Streamin example out of the box. If no, please provide us a code example based on Two Way Streaming with minimal changes to reproduce the problem, and sen using this link, we will check.

Nothing, just passing it to WebRTC library All the application examples uses auto layout. So, please prepare a code example based on Two Way Streaming (if the problem occurs when publishing) or Player (if the problem occurs when publishing) example with minimal changes to reproduce the problem...

Under the hood, WCS Android SDK uses org.webrtc.Camera1Capturer to capture frames from selected camera (and org.webrtc.Camera1Enumerator to select a camera), then uses networking part of org.webrtc to send data to server. If you could implement your custom CameraCapturer, and we could allow...

Good day. You can use REST hook /StreamStatusEvent to get FAILED status of the original stream on server. When FAILED status is detected, backend should send signaling message to frontend. On receiving this message, frontend should request HLS playlist of the backup stream. To speed up...

Добрый день. В этом классе меняется только размещение картинок в микшере, сами кадры при этом никак не изменяются. Для обработки декодированных кадров необходимо реализовать другой класс. Отметим, что, в отличие от попиксельного изменения кадров, изменение аудиосоставляющей может привести к...

Yes. You have to test a channel, adopt bitrate/resolution to channel quality, then use adopted parameters in mobile application (you can add application settings rather than hardcode them). Server itself works out of the box. But there is a channel between client and server. Channels are...

Good day. We checked your server settings. First, you must update WCS to 5.2.660 or later. In 5.2.259 that you're using now, tcp6 ports can be listened, but IPv6 WebRTC candidates are not supported yet: cd /usr/local/FlashphonerWebCallServer ./webcallserver stop ./webcallserver update After...

Hello, Configs in conf and .sh-scripts in bin will not be changed in update. For a backup, copy the directory, or the modified configs and wss.jks.

Try disabling automatic reconnect in OBS (in Advanced settings). Reproduced OBS hang-up on attempt to stop streaming when connection to WCS is restored after OBS loses connection and "reconnects" when RTMP port is closed on WCS. Created internal ticket WCS-2804 - will notify when there is an...

You can submit report using this link. We will check logs. How to create report cd /usr/local/FlashphonerWebCallServer/tools sudo ./report.sh --sysinfo --conf --tar https://docs.flashphoner.com/display/WCS52EN/Preparing+an+error+report

Hello Please check official docs Case 1 - Publish webrtc and playback RTMP 1. Publish WebRTC stream to WCS server. https://docs.flashphoner.com/display/WCS52EN/From+a+web+camera+in+a+browser+via+WebRTC 2. Playback RTMP stream from WCS server...