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Please try to re-encode file like this ffmpeg -re -i source.mp4 -acodec copy -vcodec h264 -preset fast -profile:v main -g 50 -filter:v fps=25 -bf 0 -crf 23 -maxrate 4M -bufsize 2M destination.mp4

Проблема на стороне сервера, работаем над ней в тикете WCS-3669, о прогрессе отпишем здесь

Still the issue is on Instagram side: The connection is dropping at ingest point server, so you should ask Instagram support about the issue.

Hello, In the log, RTMP connection is reset, therefore the pushed stream is stopped. Tried pushing to Instagram with the latest WCS build: sometimes connection is closed, but most times it is successful, and push is working. Please verify the stream key is correct (it starts from "rtmp/" in the...

In this case, the VOD file is transcoded. And in this case the mixer output stream content is replaced by VOD file content So the VOD file content seems the main reason of the issue. No. We need a file sample to reproduce it. Please try to encode the file without B-frames, for example ffmpeg...

По основной проблеме данной темы. В примере Screen Sharing констрейнты публикации выставляются равными размерам экрана (GitHub): VideoConstraints videoConstraints = new VideoConstraints(); DisplayMetrics metrics =...

По приведенному Вами логу, краш происходит при попытке остановки сервиса при завершении публикации экрана. К сожалению, проблема не воспроизводится на имеющихся устройствах с Android 10, 11, 12. Пожалуйста, сбросьте телефон к заводским настройкам, установите приложение и дайте необходимые...

Уточните, пожалуйста: - воспроизводится ли проблема, если публиковать поток из примера Android SDK 1.1 screen-sharing-debug.apk из архива сборки 1.1.0.59? - воспроизводится ли проблема, если обновить OneUI до 4.1 (Android 12)? Если проблема не воспроизводится в примере, модифицируйте код примера...

Good day. SIP calls recording is supported only for SIP as stream or SIP as RTMP functions. If you're using WebRTC-SIP gateway, the only way to record is to setup call recording at SIP PBX side.

Good day. SIP calls recording is supported only for SIP as stream or SIP as RTMP functions. If you're using WebRTC-SIP gateway, the only way to record is to setup call recording at SIP PBX side.

Please check the following: 1. Start a VOD stream (do not inject it anywhere) 2. Push the VOD stream to Wowza as RTMP 3. Is the VOD stream playing correctly from Wowza via RTMP? If the VOD stream is choppy, please provide the file sample using this form (if the file size is near 30 M or more...

Samsung A53 не бюджетная модель, производитель относит ее к среднему сегменту. Кроме того, оболочка OneUI на данном устройстве может быть даже более свежей, чем на S10+, поэтому просим уточнить версию OneUI и версию Android, на которой тестировали. Повтроно просим уточнить, в каком именно кейсе...

Добрый день. При трансляции экрана в примере Android SDK 1.1 Screen Sharing из последней сборки 1.1.0.59 на устройстве Samsung S10+ соотношение сторон соответствует экрану телефона Уточните, пожалуйста, какой именно кейс Вы тестируете: - публикация экрана из мобильного приложения...

You can disable or enable looping for a certain VOD file: VOD looping on demand The case seems a very complex: Mixer+Inject->RTMP->HLS, so it should be split to an elementary cases. Please answer the questions we asked in this post.

We checked the RTSP stream playback on your server. Seems the RTSP source occasionally stops to send a video traffic: In this case, in server log RTP ACTIVITY EVENT DETECTED message is displayed, and then the RTSP stream is stopped with the message Stopped by publisher stop or RtpsAgent...

The stream is playing on our test server with latest WCS build 5.2.1561 with the option rtsp_interleaved_mode=false But the stream has unstable bitrate and occasional freezes because non-interleaved RTSP uses UDP at transport layer. That is the possible reason for the stream to stop. The message...

Yes. We will check an RTSP stream capturing and server configuration.

We still cannot play in VLC the RTSP URL provided earlier. Please provide an actual RTSP URL and SSH access to your server instance using this form.

You can use command line to import an SSL certificate: Import SSL certificate using keytool Use the instance Id as admin password to login: WCS deployment from AWS Marketplace image with custom settings, p.11

Good day. Please try to split a case into some atomic tasks: - is the mixer stream playing correctly form WCS via WebRTC without injection? - is the mixer stream playing correctly form WCS via WebRTC after injection? - is the mixer stream pushed to Wowza playing correctly via RTMP from Wowza...