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You don't need to set user and password to websocket URL if you do not use a basic authentication on nginx.

Seems like nginx failed on startup: Please see here for possible fix.

The default value is 200. It should be enough for testing purposes. Try also to increase request retries count (3 by default): rest_client_request_retry_count=10

Good day. This means the internal backend handling REST hooks (RoomApi uses them) does not respond on some REST query. Please try to increase the parameter limiting a maximum simultaneous REST connections: rest_max_connections=2000

Поле sipMessageRaw в настоящее время не передается в REST хуках /call (оповещение бэкенда об исходящем вызове), /OnCallEvent (оповешение бкенда о входящем вызове) и /CallStatusEvent (оповещение бэкенда о текущем статусе звонка). Используйте информацию, доступную в этих REST хуках.

Good day. You need a CDN (see this post) with at least 15 edge servers like mentioned above

Добрый день. В сборке 5.2.1933 мы добавили более подробное логирование старта записи. Рекомендуем обновить сервер до данной сборки. В случае, если проблема со стартом записи снова воспроизведется, дополнительные логи помогут понять причину.

You have sent access to Asterisk servers. But all setup should be done at WCS side by two possible ways: 1. Set the needed outbound proxy in frontend Javascript code (Phone Min example) var outboundProxy; if (window.location.hostname === "pbx1web.xposeurope.eu") {...

This is the application level task to monitor a network state, not SDK level. Please use ConnectivityManager to register a callback to monitor a network events, and call session.disconnect() in onLost event handler: private ConnectivityManager.NetworkCallback networkCallback = new...

We checked the server. There are the following issues: 1. You have not opened the port 443 on the server, so connections to nginx reverse proxy cannot be established. Please open the port 443/tcp. 2. There is no /etc/passwd/.htaccess file on the server. Please create the file and add user and...

Good day. You've sent an access to one server, IP address for second server and SIP credentials. Please clarify what exactly you want us to do. Also, to test a SIP call we need two SIP accounts: caller and callee.

We testing the feature. Please be patient.

We tested on demo server with settings mixer_display_stream_name=true mixer_decode_stream_name=true mixer_text_font=Dejavu Sans Publishing stream with encoded characters and automatically adding it to mixer: Playing the mixer stream Maybe your font have no o with umlaut character. WCS does...

Good day. Please check your frontend code. If resolution constraints are set, a most browsers will publish WebRTC stream with resolution not higher than constraints define. If no resolution constraints set, WebSDK sets 320x240 as default. So looks like you've set 1920x1080 at client side...

Please copy the full nginx configuration example from this page: Reverse proxy setup with basic authentication for Websocket Replace the content of nginx.conf file, then change SSL certificate and key paths, domain and WCS IP address in proxy_pass directive. If nginx still does not start...

You can set the mixer font by the following parameter in flashphoner.properties file mixer_text_font=Dejavu Sans or in offline-mixer.json for multirecording mixer { ..., "mixerTextFont": "Dejavu Sans" } Please note that the font should be installed in system and should be in the X11 fonts...

Please try to switch to TCP transport as we recommended above ice_tcp_transport=true

Good day. No, it is impossible without user clicking a button or some other user interaction with page. This is a most browsers security policy which cannot be abandoned.

You have commented server in the head of this block, but left the block uncommented. Either uncomment server or comment all this block.

Good day. Use encodeURIComponent() method to encode the stream name at client side.