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Please provide SSH access to the server using this form, we will check.

Добрый день. В сборке 5.2.1009 добавлена возможность описания собственных вариантов размещения картинок в микшере при помощи языка разметки на основе XML. Разработка собственного Java класса при этом не требуется. Подробности здесь.

We prepared player.js file for Embed Player with playback restart. Restart parameters are configured with the following variables in the script: var restartTimeout = 3000; //ms var restartMaxTimes = 100; //will try to restart playback for 5 minutes So by default it will try to play a stream...

Добрый день. Данный тип реализации микшера устарел, его использование не рекомендуется. Realtime микшер отличается тем, что при проблемах с входящим потоком не ждет его восстановления, а продолжает микширование, в то время как микшер с mixer_realtime=false или...

Please clarify how do you play the stream in VLC? Are gstreamer and VLC launched on the same host? RTMP is based on TCP, so 2-3 seconds latency is normal if you publish an RTMP stream to server and play it via WebRTC.

You should either migrate to other datacenter (or get a new external IP by any other possible way), or contact Youtube support to resolve the ban.

Google Cloud https://flashphoner.com/google-cloud-platform-dlya-webrtc-cdn-s-balansirovkoj-i-avtomaticheskim-masshtabirovaniem/?lang=ru Amazon AWS https://flashphoner.com/cdn-s-balansirovkoj-i-avtomaticheskim-masshtabirovaniem-na-baze-amazon-web-services/?lang=ru В статьях описывается...

We tried to reproduce the issue with our demo server in Media Devices example (to easily change the constraints). It is playing correctly: Server build on demo: 5.2.998, WebSDK build 2.0.180 Please check if the issue reproducing in MediaDevice example on your server. If yes, please provide SSH...

If you want to play the stream published on WCS as RTSP, you can capture it from WCS by RTSP URL rtsp://wcs:554/streamName Please read details here Please check REST API query. When we tested your page last time, it seemed to be maiformed as we wrote above. Please also check if Youtube banned...

Please read this topic and other topics in developers.apple.com search results.

Good day. What iOS apps do you mean? We don't have any native iOS applications example to play HLS except WebView example, but WebView is a wrapper above iOS Safari browser. So if you're using some third party applications to play HLS, they supposed to support session stickines like browsers do.

About the ticket WCS-3275. Since Android SDK build 1.1.0.39 we added an ability to receive a stream status when a subscriber connects to this stream to play in StreamStatusPlaying event handler, using Stream.getAudioState() and Stream.getVideoState() methods. See example here.

If this is muting issue, it should be reproducible in MediaDevices example: publish, play, mute-unmute-mute... If not, try to reproduce this in Conference example: join, mute-unmute-mute... If issue still cannot be reproduced, the problem can be in your code. In this case, please modify...

You've provided log and dump from a different calls. Seems like a possible issue can be in SIP UDP signaling usage: in log we see that SDP received from far end seems to be cut. So you should either use the following parameter in flashphoner.properties file sip_force_tcp=true or exclude codecs...

Good day. You should use hardware capable enough to handle 1000 subscribers, and perform some tweaks. Please read this article containing server setup description, load testing methods and tests results defining a required server capacity: What kind of server do you need to run a thousand WebRTC...

Please clarify: do you use RoomApi or MCU mixer? payload member is filled only when user received outgoing mixer stream (MCU or not), and incoming stream publisher calls muteAudio(). Also, please reproduce the issue in WebSDK MCU Client example, it will be easier to debug. The stream is always...

Hello As we can see you are trying to use webrtc-as-rtmp-republishing sample. Please note. Your page webrtc-as-rtmp-republishing.html and your script webrtc-as-rtmp-republishing.js should connect to WCS server on port 8443. 1. You have to install and launch your own WCS server on a Linux...

Добрый день. Методики нагрузочного тестирования WebRTC приведены в этой статье. Также здесь приведена методика имитации видеочатов 1 на 1 без использования микшера.

We will check this

We've tested your page and get 500 Internal server error Seems like REST API query /push/startup is mailformed. Please check the query parameters in server logs.