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  1. Max

    Need start player every cut on stream

    Please provide the player.js code which you've modified using this form, we will check
  2. Max

    Need start player every cut on stream

    This is our demo player.html https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html This is demo player.js https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.js If you have a project you have own player.js or similar file on your web server...
  3. Max

    Ошибка при принятии звонка

    Добрый день. Ошибка No remote sdp available, говорит о том, что режим "Delayed Offer" не поддерживается. Это значит, что ваш PBX (Asterisk) либо MicroSIP совершает исходящий звонок по направлению MicroSIP > Asterisk > WCS > Browser, отправляет SIP INVITE запрос и в этом запросе нет SDP. Если в...
  4. Max

    Need start player every cut on stream

    Yes. This code can be applied to player.js
  5. Max

    Android SDK WebsocketNotConnectedException

    Please check if this crash related to this your topic. Also, please check if the issue is reproducing in Phone Min example, Streaming-min example and so on. If not, modify example code minimally to reproduce the issue and send the code using this from.
  6. Max

    Fail to start 2 streams on WebRTC server

    Please change to var rtmpUrl = "rtmp://localhost:1935/live/" + "rtmp_" + streamName; Also, try to publish in WebRTC as RTMP example as we shown above.
  7. Max

    Chrome resolution ramp-up

    Good day. To workaround this behaviour, since WebSDK build 2.0.180 videoContentHint option was added. Bu default, is is set to detail, this forces Chrome to keep publishing resolution. Please read details here.
  8. Max

    Fail to start 2 streams on WebRTC server

    We successfully published two WebRTC streams with republiushing as RTMP to localhost in two different browser windows simultaneoudly Please note how RTMP URL field is filled. Please also check the code you use for RTMP republishing: we noted the stream rtmp://localhost:1935/live is pulishing to...
  9. Max

    Микширование потоков. Скрытие всех потоков из микшера, кроме рабочего стола/окна приложения

    Размер окна потока участника (не desktop) не регулируется и зависит от количества потоков в микшере. Если необходимо регулировать размер, придется разработать собственный класс для размещения картинок в микшере.
  10. Max

    Fail to start 2 streams on WebRTC server

    Please provide SSH accees to the server using this form, we will check
  11. Max

    Memory leaks, сервер перестает отвечать

    Добрый день. В сборке 5.2.996 мы исправили проблему, приводившую к неравномерное загрузке процессорных потоков, выделенных для записи настройкой file_recorder_thread_pool_max_size. В свою очередь, это приводило к росту очередей записи и росту потребления памяти. Кроме того, добавлена...
  12. Max

    Fail to start 2 streams on WebRTC server

    In you're streaming to FB/Youtube/etc, rtmp_transponder_full_url=true is the recommended setting. So you have to set a fully qualified RTMP URL including an unique stream name per server.
  13. Max

    Fail to start 2 streams on WebRTC server

    You are using the option rtmp_transponder_full_url=true So you must use exact RTMP URL in WebRTC as RTMP case: session.createStream({ name: "1000003", display: localVideo, ... rtmpUrl: "rtmp://localhost:1935/live/rtmp_1000003" ... }).publish();
  14. Max

    Fail to start 2 streams on WebRTC server

    But you've mentioned above Flash mediaprovider. So please clarify: do you republish both streams? If yes, to what server? Provide REST API quiries. Also reproduce the problem, collect a report as described here and send using this form.
  15. Max

    Тормоза видео на HTTPS странице.

    Добрый день. В одной websocket сессии можно играть несколько потоков. Посмотрите, пожалуйста, код примера 2Players. Пример входит в поставку WCS, можете его протестировать, например, на нашем demo сервере https://demo.flashphoner.com/client2/examples/demo/streaming/2players/2players.html Это...
  16. Max

    Need start player every cut on stream

    Good day. Please try to restore stream playback automatically by tweakling player code as described here.
  17. Max

    Fail to start 2 streams on WebRTC server

    Good day. Please describe in details how do you publish stream to server: from browser via WebRTC, from ffmpeg, OBS or another encoder via RTMP, from third party server by RTMP URL etc. Also collect a report as described here and send using this form.
  18. Max

    RTSP - No Codecs Found

    Hello This means No supported codecs found. Your device sends MPEG2 (payload type RTP/AVP 33). This codec is not currently supported. Supported codec: H.264 (MPEG4 part 10). Requirements: https://docs.flashphoner.com/display/WCS52EN/From+an+IP+camera+via+RTSP#FromanIPcameraviaRTSP-Supportedcodecs
  19. Max

    RTSP Failure

    Hello We found issue related RTSP digest authentication The cam sends two WWW-Authenticate headers: MD5 and SHA-256 It seems SHA-256 is not supported on our end and we do not form Authorization response properly. We raised internal ticket WCS-3268. Will inform about progress. As a...
  20. Max

    Screen Share and Video on Flashphoner Conference Feature

    You should either stream every participants video separately, or use stream mixer to combine all the streams to one. Anyway, the stream can be republished to Facebook or Youtube as described here.
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