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Good day. /stream/snapshot REST query has no uri parameter, see the example /stream/snapshot and read details here.

Good day. Video is not cropped in mixer, it is mixing as is, with width and height received from publisher. But you should aware of aspect ratio. Seems like the entire screen has 16:9 aspect, and browser tab or window has 4:3 aspect. It may be scaled in this case. Also, the explicit height...

Please provide SSH access to the server and RTSP URL to test using this form. We can't help you by just a screenshot. All we can test is other RTSP links playback, and it is working on your server: If you can't provide an access, please do the following: 1. Enable client debug logs on server...

Please try to reproduce the issue in Player example https://your-wcs-instance-ip:8444/client2/examples/demo/streaming/player/player.html. If the problem is not reproducing, modify the Player example code minimally to reproduce it and send the code using this form. If the problem is reproducing...

Good day. A valid session states are: ESTABLISHED, DISCONNECTED, FAILED. See example code here Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function(session){ setStatus(session.status()); //session connected, start playback...

You should deploy a web server (Apache or nginx) with PHP module for example. Then you should make a PHP script that handles a REST hooks needed on the endpoint http://host/rest-hooks/. And, finally, you should configure WCS to call to this endpoint. Please read a step by step example which...

Good day. Please make sure you've applied all the tweaks listed here: Before moving to production and all the tweaks listed in section Preparing for testing of this article. Suppose you're publishing 640x360 1 Mbps stream. In this case, you can serve maximum 800 viewers on 1 Gbps channel (800 *...

You can use /StreamStatusEvent REST hook to check if RTSP stream becomes FAILED. Please look at REST hook implementation example that can be used as template.

Good day. You can use REST API to check if RTSP stream is running or stopped, and start to capture it again when it stopped.

Good day. We added support for separate custom Java libraries folder in build 5.2.1512. Please read details here A separate folder for custom mixer Java libraries

Good day. Please check if WCS listening all the ports needed. Server health check after start uses HTTP port 8081 by default. Use the command netstat -nlp | grep java to list all the listening ports. Also. in the log we see BindException It means server cannot bind media UDP port in range...

We've tested the layout 2_participants.desktopmix mentioned in the first post, and we can't reproduce the issue you describing: Also, the issue is not reproducing for a sample FullScreenDesktop layout Please update WCS to the latest build 5.2.1510. If the issue still persists, provide SSH...

Please see the message above. Make sure you've set two names in -N option for two participants layout.

You are adding only one stream to the mixer in this case. The -N option should include all the streams name if it is used: /mixer_layout_tool.sh /opt/mixer-layout-vert-collab -o=/tmp -N=test1-screenShare,test -n=2

Hello, Please let know whether a non-default value is specified for mixer_video_layout_desktop_key_word in flashphoner.properties. If the option has the default value (desktop), .mix (not .desktopmix) file will be used if none of the stream names contains desktop (like test1-screenShare).

Good day. Flashphoner Web SDK works only with Flashphoner Web Call Server, but not with Kamailio server. You should deploy your own WCS instance in AWS EC2 or in Digital Ocean, or on other cloud or hardware server, and test Web SDK examples with it. Please read here how WebRTC-SIP gateway...

We already working on NVIDIA GPU transcoding, it is on beta testing stage. There is the ticket WCS-3652 to prepare it for production, but we can't provide ETA yet.

Good day. There is the ticket WCS-3106 to support H265, but Chrome still does not support publishing, but playback only. So we plan to implement H265 support for RTSP streams capture first. Also, seems like Chrome 108 on Windows only plays the test videos from this page...

We can't recommend a certain SIP service provider. You can ask your SIP provider about RFC 5589 support for call transfer. Most software PBX support this specification.

По тикету WCS-3662: проблема исправлена в сборке 2.0.155 (доступна в NPM: @flashphoner/sfusdk).