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Good day. The default policy for calls is LOG when backend returns 403. So, we have to change the policy: 1. Put the file rest_client_config.json near the backend script. The file should contain at least the following: { "call" : { "clientExclude" : "", "restExclude" : ""...

Good day. You can use REST hook to authenticate user. Here is described detailed example to authorize users by domain, use this example as basis. If there is no any subscriber (viewer) for transcoder output stream, the trascoder stops by inactivity after 60 seonds by default. You can set this...

You should set your own server IP-address or domain name here: https://{your server domain name or IP address}:8444/client2/examples/demo/streaming/media_devices_manager/media_device_manager.html because this is Media Devices example URL on your server.

Use - rest-api/pull/rtmp/pull to pull the RTMP stream - rest-api/transcoder/startup { "uri": "rtmp://fms.105.net/live/rmc1", "localStreamName" :"5e0c" } to create a new stream with a local name rest-api/transcoder/startup can be used to create a transcoded version of a stream, but if the...

Try different resolution, bitrate and fps values both with UDP and TCP transport to see which would give better result. E.g., if bandwidth allows - increase bitrate and fps, and if bandwidth is low - decrease bitrate, fps and resolution. The Media Devices example can be used for the tests...

Hello, To establish Secure WebSocket connection, SSL certificates have to be imported. To use unsecure connection, use this URL: http://WCS:8081/admin/demo.html Also, the RTMP stream you are using for testing contains B-frames. There may be lags during playback.

Good day. If your WCS server and WebRTC stream source are in same datacenter, the stream quality is probably not a publishing but playback problem. So try to use WebRTC over TCP for playback, this helps to reduce channel lossess for viewers, but it may increase latency.

Good day. Please disable graphic acceleration feature in browsers and check if problem still persists. If yes, please collect a report as described here including traffic dumps separately for every browser, and send to support@flashphoner.com, we will check.

Good day. You should create a folder on your backend server, for example /var/www/html/rest-hooks assuming Apache. Then set up your web server to start index.php from this folder. In index.php get REST hook method as $api_method = array_pop(explode("/", $_SERVER['REQUEST_URI'])); So method URI...

Good day. Please clarify your case: 1. You make a call from browser using WebRTC-SIP gateway functions, and it works. 2. You make SIP as stream call to republish SIP call to another server, 3. Or you try to republish incoming SIP call as stream to another server, and it does not work? Anyway...

Good day. The WCS version in Amazon Marketplace will be updated in the coming days.

Добрый день. Используя REST-метод /vod/startup, в uri вы указываете из какого файла захватить VOD поток. Он может быть расположен локально на вашем сервере или находиться в хранилище AWS S3. Для воспроизведения локального файла (например, filename.mp4), разместите его в директории (если её нет...

Good day. Please update your server to the latest build 5.2.220 from this page, it also includes update for WebSDK. If problem still persists after update, please collect a report as described here, including traffic dump, and send it to support@flashphoner.com. Please do not modify logs and...

You can upgrade just WebSDK in client2 folder and your code. No need to update server side, this is only Safari browser problem.

Regarding keep alive, it is described in OBS related docs https://docs.flashphoner.com/display/WCS52EN/Using+OBS+Studio#UsingOBSStudio-Knownissues

Attached screenshot Settings - x264 ultrafast. Please check. Yes REST hooks works for OBS too. When you publish RTMP stream to server, REST hooks sent to flashStreamingApp URL Type CLI command: show apps -d to see actual URL for flashStreamingApp REST hook One major detail. OBS does not...

Hello There are two OBS streamers: OBS RTMP, OBS WebRTC (customized). You are using OBS RTMP. Try to change encoding settings and set x264 ultrafast preset. It should decrease latency and freezing. We have also customized OBS WebRTC for low-latency streaming. Planning to release next week.

Good day. We solved problem (internal ticket WCS-2055). There was a problem with an unsupported resolution in iOS, more detail, you can learn in the documentation. Сhanges in WebSDK you can see on GitHub. You should update WebSDK to latest build from this page as follows: 1. Download latest...

Good day. We will test a WCS with this case and write to you a little later. Write which a WCS version you are using: cat {WCS directory}/conf/WCS.version and WebSDK version: cat {WCS directoty}/client2/client.version

Good day. We make every effort to satisfy users, but forum support does not imply any SLA parameters on response time. You need at least 32 Gb of RAM for this broadcasting configuration and this processor (4 cores) is also not enough.