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Добрый день. Это известная проблема, работаем над ней в тикете WCS-3211, релиз ожидается сегодня-завтра. Проблема проявляется при публикации потока с нестандартным форматом пикселя и транскодинге такого потока на сервере. В качестве временного решения рекомендуем откатить WCS на сборку 5.2.961.

Тикет WCS-3186 находится на предварительном тестировании, релиз ожидается до конца этой недели

Yes, it is possible. You should set publishing framerate in constraints: session.createStream({ name: streamName, display: localVideo, cacheLocalResources: true, receiveVideo: false, receiveAudio: false, useCanvasMediaStream: true...

Please re-check your flow by adding console logging for example. When switching from one stream to another, you should stop the previous playback before playing new stream. Stream.stop() method always removes playing video tag. But publishing video tag may be cached by default, in this case it...

Good day. HLS can give latency up to 15 seconds by design. We are working on LL-HLS (low latency HLS) implementation in ticket WCS-2242, it is on development stage. We'll let you know when release it. Also, in the latest iOS version, you can use WebRTC both in Safari and third party browsers...

Press CTRL+C in the terminal to stop capture and write pcap file.

Try tcpdump -i ens192 port 8081 -B 10000 -w log.pcap Here ens192 is network interface available from command ifconfig. Example: [root@demo /]# ifconfig ens192: flags=4163<UP,BROADCAST,RUNNING,MULTICAST> mtu 1500 inet netmask 255.255.255.240 broadcast 188.40.244.255 inet6...

tcpdump port 8081 -B 10000 -w log.pcap or if you send REST request from local PC, just start Wireshark and capture traffic from UI

REST API can return 404 Not found if no target stream found. You can do REST query to http://host:8081/rest-api/push/startup and capture HTTP traffic by Wireshark or tcpdump. 1. Please send us captured HTTP traffic. We will check validity of HTTP request. 2. Please send output of page...

Please send a minimally modified code of this example to reproduce the issue. Just html and js in archive, no PHP, no fiddles. Seems like you never stop participant stream when playing a new one. In this case, a new video tag will be created per every new playing stream.

Please provide SSH access to the server this private form. If this is impossible please collect a report as described here including client debug logs and traffic dump collected on server. Traffic dump collection should be started before calling /rtsp/startup. Send the report archive with...

The stream you've provided is playing normally on our demo server in Player example, please check. So you should check if this stream is available from your WCS instance: - install ffmpeg - try ffplay rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov - look at ffplay output

Please check if RTSP stream can be played using ffmpeg/ffplay/VLC. If yes, please provide us RTSP link (should be available 24/7) to check the stream using this form. If you cannot provide the link available from outside, please provide SSH access to the server and RTSP link in private network...

Good day. Is the issue reproducing with sample code from this page? If not, please modify the code minimally to reproduce the issue and send the code modified using this private form

Добрый день. При тестах на сборке WCS 5.2.968 и сборке WebSDK 2.0.170, проблема не воспроизводится в Firefox 89.1.1 на следующих устройствах: Nokia 5 (Android 9), Google Pixel 3 (Android 11), Xiaomi Redmi 9 (Android 10, MIUI 12.0.2) и воспроизводится только при публикации H264 на Samsung S10+...

Добрый день. В сборке 5.2.968 добавлена возможность вернуть собственное сообщение об ошибке в теле ответа. Подробнее здесь.

Good day. First, please check if you are using Android SDK 1.0 on Android 5-7 devices only. On Android 8 and higher, use Android SDK 1.1 (latest build 1.1.0.27). Also, 1.0.1.35 is too old. Please update to latest Android SDK 1.0 build 1.0.1.77.

Good day. This seems like browser issue definitely. You can try one of the ways to prevent caching or to clear cache on page load.

You should use constraints session.createStream({ name: streamName, display: localVideo, ... constraints: { audio: true, video: { width: 640, height: 480 } }...

We made some tests on Windows 10 PC (i5-8250U 4 physical cores, 20 Mb RAM) playing 25 streams (360p 25 fps, bitrate 1,5-3 Mbps) in 3 Chrome tabs. During the tests CPU load were in range 40-50 % (25% GPU load), without noticeable freezes or stream visual degradation. So WebRTC playback capacity...