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Yes, any stream published to WCS server is available to play via WebRTC, including mobile SDKs Yes, if a platform provides an RTMP URL to play Yes. Please look at the following examples: Video Conference Flash Video Chat Android Video Conference iOS Video Conference

Hello >Is there a new WCS release also required? No. This fix is Web SDK only. >will the video chat demo sample need any change? You can track all changes on our github with comment #WCS-3445 https://github.com/flashphoner/flashphoner_client/commits/wcs_api-2.0 Therefore you can see all...

You can try to use Adobe AIR on mobile devices. Flash example applications sources are still available with WebSDK, so you can build them and try with Adobe AIR. If this works, you can publish RTMP streams from mobile devices and play them from desktop as WebRTC using Javascript API.

In WebSDK build 2.0.216, it is possible to pass participant stream playback options. Use this to disable automatic stream audio unmuting in Android Edge browser and to prevent a participant stream freeze. Then, user must click a button to unmute audio var options = { unmutePlayOnStart...

Good day. Unfortunately, Flash is dead and not supported in all the modern browsers and operating systems. But WCS still supports RTMP publishing and playback. You can even build WebSDK from sources including Flash support if you find a browser which still supports Flash. So we recommend to use...

Здраствуйте. Из коробки, такой функции нет. Есть возможность перехвата и обработки сигнала, к которой вы можете подключить свои алгоритмы (фильтры) изменения голоса. https://docs.flashphoner.com/display/WCS52EN/Server+audio+processing На входе сигнал PCM16

It will be released tomorrow

Good day. Please provide an access to the server, SIP PBX credentials and two SIP accounts to make a test call between them using this form.

Good day. Please check the following logs: 1. Server log around the time when server stopped /usr/local/FlashphonerWebCallServer/logs/server_logs 2. System messages log /var/log/messages 3. Check if there is a log file /usr/local/FlashphonerWebCallServer/logs/errorPID.log, where PID is process...

Добрый день. В сборке WebSDK 2.0.215 добавлена возможность определять параметры логирования не только для приложения в целом, но и для сессии, потока или звонка. При этом на каждом из уровней используется свой экземпляр объекта логирования. Пример приведен здесь: Определение параметров...

Добрый день. Возможная причина в Вашей последовательности операций (flow). По логам (в том числе и предоставленным ранее) она выглядит так: 1. Ведущий публикует поток 2. По REST API /mixer/startup создается микшер 3. По REST API /mixer/add поток ведущего добавляется в микшер 4. К потоку микшера...

Добрый день. Если речь идет о захвате потока с камеры на WCS, то поддерживается передача имени и пароля в RTSP URL, при этом служебные символы должны быть экранированы: Аутентификация при захвате потока Если речь идет о проигрывании захваченного потока по WebRTC, то есть возможность стартовать...

Мы проверили Ваш сервер. Если публиковать поток в микшер (например, fptest#fpmixertest), на подключая зрителей к микшеру, то запись успешно создается, и после остановки микшера по таймауту отрабатывает скрипт on_record_hook.sh, в котором запись копируется в AWS S3 хранилище, после чего удаляется...

So, you publish video from canvas like Canvas Streaming or Stream Filter example. In this case, please chek if the stream published may be played from the server (before adding to mixer). Also, please check the stream metrics using /stream/find REST API: POST /rest-api/stream/find HTTP/1.1...

Good day. Do you play the file as VOD from S3 storage and then add this to mixer? Or do you use a custom mixer layout implementation? Please clarify the case step by step to reproduce. Please also note that that video may freeze when playing VOD from S3 storage if server and file storage are in...

Not yet. Please be patient.

Usually, mp4.tmp is saved as mp4 after recording is stopped either via REST API or by stopping a stream itself. The saving may fail due to low disk space or if WCS was eventually stopped or restarted while recording. Anyway, mp4.tmp cannot be converted manually because it has no MP4 metadata and...

Hello .tmp means file is being recorded at this moment You have to stop publishing stream to finalize file. mp4 header will be written into the beginning of file upon finalizing. So you can play or convert stream after finalization only.

May be this year. Definitely, after WCS-3420 (CallKit usage example) resolution.

Hello, Imported the certificates on a test server to check using openssl: openssl s_client -connect <domain>:8444 Gives these errors 140607821154192:error:0407006A:rsa routines:RSA_padding_check_PKCS1_type_1:block type is not 01:rsa_pk1.c:103: 140607821154192:error:04067072:rsa...