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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

Hello Each running and activated server sends pings. 3-4 pings per hour. If at least one ping sent within the hour, the billing system will calculate this as a unit. Example1 10:01 AM - Server A sends ping 10:15 AM - Server B sends ping 10:30 AM - Server A was crashed 10:45 AM - Server B...

Hello Could you provide the RTSP link? We will check. You can also send RTSP link via the private form.

Good day. You can implement a custom Java class to intercept every frame and crop it programmatically: Decoded frames interception and handling Another way is to use stream mixer with a special cropping layout, but this only crops a central area of the stream picture: Crop around center zero...

Good day. A typical stream flow is: 1. PENDING - WebRTC connection is establishing, no media traffic yet, stream codecs is unknown 2. PUBLISHING - WebRTC connection is established, media traffic is sent from client, stream codec can be detected 3. UNPUBLISHED or FAILED - streaming is stopped by...

Привет. Мы не тестировали запуск под Windows 11 в Docker. Потребуется некоторое время чтобы проверить запуск. Для Windows обычно рекомендуем запуск WSL

We've updated demo server to the latest build 5.2.1131 and WebSDK 2.0.212. In this case, the problem is reproducing randomly in Chrome 98.0.4758.80 on MacOS 12.2. Seems like the issue is ih hardware H264 encoder parameters. There are the following workarounds: 1. Publish VP8 codec instead of...

Good day Please reproduce the issue in Two Way Streaming example. If it's reproducing please provide Chrome Dev Tools console screenshot.

Please clarify: what exactly result you've got on every step (1, 2, 3, 4)? Also check if examples are built correctly with Gradle from command line.

Yes, ICE connection is establishing only for WebRTC publishing/playback. And seems this is broken yet on some of Samsung devices (but on good old flagships like S10/S10+ ICE works correctly both in Chrome and Samsung browser). So, we have to wait for Samsung fix and use Firefox as workaround.

Thank you for investigation. Please, check if the problem is reproducing in latest Samsung browser build (they may fix it in proprietary browser). If this is not fixed yet, try also to publish VP8 in Media Devices example: If codec stripping does not help, seems like Firefox is the only...