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Good day. Please use Player example to test: https://demo.flashphoner.com:8888/client2/examples/demo/streaming/player/player.html If your RTSP stream is not playing in this example, please provide RTSP URL to test (the camera should work and should be available 24/7) using this form.

Добрый день. Это известная проблема, возникает при увеличении размера SDP при пересогласованиях, например, при добавлении/удалении дорожек. Работаем над ней в тикете WCS-3669. О прогрессе сообщим.

You can change resolution at the example page: function getConstraints() { var constraints = { audio: {deviceId: {exact: $('#micList').find(":selected").val()}}, video: { deviceId: {exact: $('#cameraList').find(":selected").val()}, width...

Seems like channel bandwidth between server and publisher/player browsers in not enough to publish a desired resolution. Use lower resolution/bitrate and/or try to switch to TCP transport. Also check a client network conditions: do not use mobile 2G/3G, use 4G/LTE or WiFi or a wired connection.

Mixer fails because of compact_media_port_usage=true. Please remove the option from flashphoner.properties and restart WCS.

Please set media port range to default media_port_from=31001 media_port_to=32000 Disable RTP paced sender (we do not resommend to use it in production rtp_paced_sender=false Revert media_ports_auditor_interval to default (comment or remove from config) Then restart WCS. If this does not...

The same credentials with the same result If you cannot provide an access, please restart WCS, reproduce the problem and collect the full report: Getting logs with report.sh script, then send the archive using this form

Unfortunately, the server rejects SSH connection with Permission denied, please try again. Seems like either login or password is incorrect.

Please provide SSH access to the server using this form.

Yes. A local internet service provider (or mobile network operator if used). So it's between the client and the server. A provider can use deep packet inspection (DPI) technology to filter traffic at UDP packets level. So client cannot establish TURN connection because network packets are...

Hello, The demo server is working, RTSP can be played now.

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream Also, H265 MPEG-TS stream may be published: H265 publishing Note that H265 requires transcoding to play it as WebRTC. Chrome does not play H265 via WebRTC, only from media...

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream Also, H265 MPEG-TS stream may be published: H265 publishing

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream

Добрый день. По поддержке RTSP камер с кодеком H265: в сборке 5.2.1579 добавлена возможность захвата RTSP потока, публикуемого камерой в кодеке H265. Подробнее здесь: Захват H265 RTSP потока

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream

UDP TURN traffic may be blocked by provider, for example, by packet signature. We recommend to use TCP for relay.

Good day. Seems like server cannot bind UDP port when mixer media session is created: Please check if ports 30000-33000/udp are not used by any other application on the server.

Выглядит так, что от SIP стороны в INVITE SDP приходят кодеки speex 8000 и speex 16000, которые Chrome не может использовать одновременно Поэтому браузер не может применить SDP, которое предлагается сервером. В наших тестах (звонок с софтфона со следующими кодеками: Speex 8000, Speex 16000, GSM...

Пришлите пожалуйста также лог консоли в Chrome браузере. Developer Tools > Network > domain Среди этих сообщений должен быть SDP конфиг.