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Good day. We have checked logs, but the reason for ICE failure by timeout still is not clean. Please provide us SSH access to the server and SIP account details for test call, we will check inplace. If acces cannot be provided, please do the following: 1. Check if SIP call to local Asterisk...

Добрый день. Да, Вы правильно поняли логику работу нагрузочного теста микшеров. Добавим к этому, что по истечении intervalInSecons микшеры уничтожаются и создаются заново, тест продолжается до вызова /mixer/test/stop, либо до остановки сервера. Поэтому рекомендуется задать шаблон имени файла...

Hello, The call in the log "Failed by ICE error". Please check firewall and routing settings and verify that the UDP ports are reachable. In case of further investigation, please send additional information (with server config and ifconfig result) per this instruction.

Hello, MP4 format is supported for recording H.264 video stream. WebRTC audio-only stream also can be recorded as MP4. The format is not configurable.

Good day. You have provided recording examples done not in the same time for mixer incoming stream (obsrtmp) and mixer outgoing stream (mixer1). It seems like mixer drops incoming frames, and log messages in the post above show this. So we suppose to enable incoming stream buffering by enabling...

You should either update with commands cd /usr/local/FlashphonerWebCallServer/bin webcallserver stop webcallserver update webcallserver start or update by downloading archive and installing over previous version cd ~ curl -sL -O...

The RTSP authentication issue has been fixed in WCS v. 5.2.470.

Good day. By default, WebSDK normalizes stream publishing resolution constraints set in Safari browser. In this case, if width or height is not set, or equal to 0, then the picture resolution is forced to 320x240 or 640x480 like this: ... if (!width || !height) {...

Hello, It is necessary to confirm quality of the stream pulled to the mixer server. Please try it first without creating mixer. 1. In case of pulling as WebRTC (/rest-api/pull/pull), try using TCP: add this setting to flashphoner.properties on the mixer server ice_tcp_transport=true and...

The issue has been reproduced with the latest received credentials to the stream - submitted internal ticket WCS-2456, will notify when there is an update.

Добрый день. Создали тикет WCS-2454, о результатах сообщим здесь.

Good day. This is a known issue, any non-latin (cyrillic, korean, japanese, extended latin...) characters should be URI-encoded while sending text message: addMessage(participant.name(), encodeURIComponent(message)); and decoded while receiving it: ... }).on(ROOM_EVENT.MESSAGE...

defaultApp is one of the server-side applications. An application stores a REST URL of back-end web server ("yourhost" in the URL example above). By default, it is configured to work out of the box without additional back-end setup. The URL can be changed (using the "update app" CLI command) to...

In the received log, the failure was due to other reason: "Back-end has returned 401 Unauthorized on connect". The server has disable_rest_auth=false in the config - REST hooks should be used for authorization.

The update issue is fixed in build 5.2.239. We recommend you to update manually as described here to latest build from this page.

Please check if WebRTC stream publishing and playback is slow too using Two Way Streaming example. If yes, please collect report as described here, add WCS version you use and send it to support@flashphoner.com, we will check. If Two Way Streaming works normally, collect report for Conference...

Good day. Please clarify do you use Flash-based conference (Flash Video Chat example) or WebRTC (Conference example)? Note that Flash support in browsers supposed to be dropped this year. Also note that WCS RoomAPI is just a wrapper on WebRTC stream publishing and playback functions with text...

Good day. Please check if your credentials contain any special characters like this: rtsp://username:p@ssword@myIP:100081/0 In this case, the special character should be escaped like this: rtsp://username:p%40ssword@myIP:10081/0 If this tip does not help, please collect a report as described...

Good day. To use autoplay, you should start playback with muted audio, then user should unmute with volume control. We cannot reproduce freeze on autoplay in iOS 13.3 using latest WebSDK. Do you use playFirstVideo function like Player example? function playBtnClick() { if (validateForm())...

Good day. Is the issue reproduced in Player example with autoplay=true parameter (https://WCS:8444/client2/examples/demo/streaming/player/player.html?streamName=stream&autoplay=true) or in your own code? In last case, please provide us minimal Player code modification to reproduce.