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Good day. Please clarify your case: do you publishing a stream from third party RTMP encoder and playing as WebRTC? Do you want to authorize stream viewers? If yes, please look at /connect and /playStream REST hooks. There is a step-by-step example to implement /connect REST hook which can be a...

We support both Objective С and Swift, but GPUImage3 is Swift only framework. So we have to stay with GPUImage1.

Hello We have customers who do similar things such as face detection, beautification, etc. We have implemented support of GPU Image Library for this case. However we didn't use any filters just very simple custom filters for demonstration purposes. You would need to search more about GPUImage...

Filters preview is available in example applications, so you have to build them to test. In Web SDK, all filters code seems trivial: GitHub In Android SDK and iOS SDK, GPU Image library is used, so you can refer to its documentation to implement a custom filters. Note that GPU Image has a...

Now it seems like the console example is working: Please do not forget to enable CORS in the extension: toggle this picture if it is grayed

You can try to disable stats at all using HTTP interfaces access restriction: by default, the setting is http_enable_paths=rest,action,admin,shared,client,client_records,embed_player,empty,health-check Removing action from the list should disable statisctic interface. A minimal set of...

Please use this extension to enable/disable CORS in Chrome browser: Allow CORS: Allow-Control-Allow-Origin This page returns empty response: Please check if WCS is running on this address and listens the port 9091. Make sure this is not a load balancer address. If nofing helps, please provide...

Good day. Yes, you can use filters during live streaming. Please see the following examples: Web SDK: Stream Filter iOS SDK: iOS GPUImageDemo Swift Android SDK: Android GPUImage or Android Camera Manager

You can increase media ports range using the following settings in flashphoner.properties file: media_port_from=20001 media_port_to=40000 WCS needs to be restarted to apply changes.

About HLS latency. You can use Low Latency HLS since WCS build 5.2.1345 and newer: Recommended settings for LL HLS playback. In this case the latency may be no more than 10 seconds (usually 1-4 seconds which is normal for TCP based protocols). Note that HLS player must support LL HLS, for...

Hello Autoplay policy is being changed in browsers. You can read more here: https://docs.flashphoner.com/display/WCS52EN/In+a+browser+via+WebRTC#InabrowserviaWebRTC-Autoplayissuesindifferentbrowsers 1. How can we reproduce this issue? Please provide screenshots. 2. If we open page 10 times...

Добрый день. В сборке 5.2.1349 добавлена возможность динамически добавлять и изменять водяной знак, не останавливая транскодер или микшер. Подробности смотрите здесь: Динамическое добавление и изменение водяного знака в поток транскодера Динамическое добавление и изменение водяного знака в...

The best way to exclude codecs is to strip them on client side: session.createStream({ streamName: "test", display: remoteVideo, ... stripCodecs: "VP8" }).on(STREAM_STATUS.PLAYING, function (stream) { ... }).play(); You can test it in Media Devices example...

There is no known side effects if you just remove codecs from codecs= Anyway we recommend exclude instead of remove for better differentiation by feature-sets. codecs_exclude_cdn codecs_exclude_sfu codecs_exclude_sip codecs_exclude_sip_rtmp codecs_exclude_streaming

You should better exclude vp8 codec then removing it. codecs_exclude_streaming = vp8 Please note. If vp8 codec is excluded, some devices will not be able to play the stream and you will have error playback reports. Generally some Android Devices (Android Apps and some Android Browsers) do not...

Hello Are you sure that video.js can play RTMP streams? Adobe Flash Player is not currently supported in most of browser versions. As we know video.js uses swf (Adobe Flash) binary for RTMP playback. Make sure your browser supports Flash (maybe you are using quite old browser version with...

Seems you've set Java heap to 16 Gb. Now there are 7 published stream on server and ~350 viewers. This load seems to be acceptable, but you have a lot of subscribers playing VP8 codec. This means sream transcoding H264->VP8. Also, there is audio transcoding AAC->Opus. So now you have ~20% CPU...

We checked the report. It seems like you have published more streams than theserver can handle, or more subscribers. Please read this article about server configuration and testing under high load: What kind of server do you need to run a thousand WebRTC streams? Also, seems like you're using...

Place the archive to some cloud drive (Google Drive, OneDrive etc) with public access and send the link in Comment field of the report form Please get the public key on this page: Providing support engineer access to your server

Please do not use Skype, direct email etc. You should send the report archive or SSH access using this form only (or click the Report button at the top of this page) Also please make sure you're using the latest WCS build 5.2.1346.