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Seems like Huawei Cloud exposes AWS-like API. In this case, admin password is set to instanceId, which can be retrieved as follows curl http://169.254.169.254/latest/meta-data/instance-id Also, it can be displayed somewhere in cloud console. Example fow AWS:

We've checked your server. Seems like license server connection was broken occasionally. We restarted the server systemctl restart webcallserver then it works correctly:

Good day. We have not tested WCS in Huawei Elastic Cloud yet. Please provide SSH access to the instance using this form.

Good day. Seem like some streams are publishing with video. To chek how streams are publishing on server side, please set up Prometheus and Grafana to collect server monitoring data as described in this article 10 Important WebRTC Streaming Metrics and Configuring Prometheus + Grafana...

Возможно, ответ есть в документации по ReplayKit.

Please download latest build 5.2.1133, then unpack it. The default database.yml file is in FlashphonerWebCallServer-5.2.1133/server/conf folder.

We checked your server. Seems you have an instance in DO, with build 5.2.780 out-of-the-box So the examples opened directly from server are working, but a newer examples may not. We recommend you to update WCS (with WebSDK included) to the latest build 5.2.1133 as described here.

Hello What is your server version? Example: Our demo server: https://demo.flashphoner.com Server: 5.2.1131 Web SDK: 2.0.212 view-source:https://demo.flashphoner.com/client2/examples/demo/streaming/video-chat/video-chat.html flashphoner-room-api.js used You can download latest server...

Good day. Screen sharing with a native screen resolution (1080p and above) requires a good channel quality like for 4K broadcasting: 20 Mbps bandwidth with a zero loss. If the user has such a good channel, you can use UDP transport which reduces a latency. Otherwise, you have to use TCP...

Good day. Seem like your server does not respond on port 8443. Please provide SSH access to the server using this private form.

Добрый день. Чтобы определить, начался ли захват экрана, можно использовать событие UIScreen.capturedDidChangeNotification. Посмотрите этот код: NotificationCenter.default.addObserver(forName: UIScreen.capturedDidChangeNotification, object: UIScreen.main, queue: OperationQueue.main) {...

Hello Could you share recorded file and RTSP URL You can do that privately using this form or post this information right here Recorded file can be uploaded on google drive or any storage resource. We need just a download link. How do you record the stream? Using REST API or settings? Please...

No, there is no such example because it is a little more complex task. If you've set the server option record_rtsp_streams=true and play an RTSP stream in Player example, you should use REST API /recorder/find_all POST /rest-api/recorder/find_all HTTP/1.1 HOST: wcs:8081 content-type...

Hello, By default, records are saved to /usr/local/FlashphonerWebCallServer/records. Record file name template is specified by stream_record_policy_template. The template can be changed to include stream name (':' and '/' will be replaced with '_'). The parameters can be configured in...

Thanks for clarification @tharun_n so you have maximum 225 Mbps bandwidth (with 10% reserved) for configuration you use. The bandwidth is enough for 200 streams of 1 Mbps each, or 100 incoming RTSP streams and 100 outgoing WebRTC streams (one per each subscriber). This is more than a maximum...

Please look at Media Device example source code. To allow user to choose the microphone, you should: 1. Iterate through the all audio input devices code Flashphoner.getMediaDevices(null, true).then(function (list) { list.audio.forEach(function (device) { var audio =...

Hello Take a look at this https://flashphoner.com/how-many-rtsp-cams-can-i-connect-to-the-webrtc-server/ It is a load test for 1 CPU, 12 cores, 24 threads; 96 Gb RAM; Result: 1000 RTSP streams You have to carry out your own load tests to see the capacity of your instance. Please be noted...

Good day. Please check if Mic selector is working in Media Devices example If the microphone you need is not in list, then it probably is not available for browser. The system audio can be captured in Chromium browser only when capturing a screen

Try to escape character @ in the password Replace rtsp://admin:pass@word@localhost:554/live.sdp With rtsp://admin:pass%40word@localhost:554/live.sdp Works in our test. Please note. The "admin" and "pass@word" should be URL encoded before sending to the player. Or you can modify embed...

You can set valid RTSP URL and add your custom URL as a text comment