Recent content by Max

  1. Max

    Installation problems on Ubuntu 24.04, Java 17, WCS 5.2.1930

    Perhaps a newer WCS builds requires non-empty REST hook response, but older builds may be not so strict.
  2. Max

    Installation problems on Ubuntu 24.04, Java 17, WCS 5.2.1930

    We checked the server. There are the following problems in configuration: 1. Non-critical: there are no such parameters anymore in flashphoner.properties like sip_enabled=true rtp_force_synchronization=true Please remove them 2. Critical: seems like your SIP PBX does not respond to REGISTER...
  3. Max

    Installation problems on Ubuntu 24.04, Java 17, WCS 5.2.1930

    Good day. Please make sure you've done a clean WCS installation, not directly update from 5.0 or earlier versions. Also please make sure you do not unpack an installation archive directly to /usr/local folder. If this recommendations do not help, please provide SSH access to the server using...
  4. Max

    Update docker image (recent OS and JDK)

    Please be careful: Docker is easy to deploy but it's very hard to debug WebRTC in Docker in production under high load. Read this article for example: How to use Docker with WebRTC in production
  5. Max

    Update docker image (recent OS and JDK)

    Good day. The official Docker image is updated: - Ubuntu 24.04 - OpenJDK 21.0.2 - WCS 5.3 Please read the updated docs: WCS in Docker
  6. Max

    Error republishing RTMP stream

    Good day. We cannot reproduce the issue with build 5.2.2292. Please reproduce at your environment, collect a report as described and send using this form.
  7. Max

    Moving WCS Mixer Recording file to other location

    Good day. The following parameter mp4_container_write_header_on_fly=true may help. In this case, MP4 header and MP4 data will be written to a separate files then concatenated when recording is finished (by stream stopping or by REST API query). This would be much faster. The record hook...
  8. Max

    Проигрывание HLS и снепшоты

    Добрый день Эти две проблемы связаны с периодичностью поступления ключевых фреймов от RTSP камеры. Если есть такая возможность, рекомендуем выставить на стороне камеры отправку ключевого фрейма каждые 2 секунды (если, например, у камеры есть опция keyframe interval наподобие OBS). Если такой...
  9. Max

    DigiEye's streams cant be played

    Good day. We checked the report sent via e-mail. There are no any issues in the server logs: the RTSP stream is captured successfully, then (after 40 seconds) the client stops stream playback There is no the client log in the report. Also, there are no SDR logs. Next time please collect a full...
  10. Max

    Переподключениче стрима

    Для WebSDK на стороне клиентского кода можно реализовать автоматическое переподключение зрителя: Автоматическое восстановление воспроизведения потока, Streaming Auto Restore В этом случае заглушку с отображением статуса реконнекта необходимо также отрисовать на клиенте.
  11. Max

    Переподключениче стрима

    Здравствуйте Под "Стрим не прерывался" мы понимаем следующее поведение: 1. User1 заблокировал экран устройства, устройство закрыло соединение с сервером. 2. В это время, зрители продолжают видеть черный экран заглушки. 3. Зрители дожидаются когда User1 восстановит стрим, без отключения. 4...
  12. Max

    Организация работы CDN

    Добрый день. Нет, это неверно. Когда какой-либо сервер в CDN запускается, ему нужна точка входа: адрес узла, с которым сигналинговое соединение будет установлено первым. После того, как это соединение установлено, узел получает описание текущего стейта CDN со списком всех активных узлов, и...
  13. Max

    IOS 18.4.1 webrtc video and audio out of sync

    Good day. Since build 5.2.2269 we've added the userAgent parameter to identify any browser which is needed to be fixed. Please update and add the following parameter to flashphoner.properties quick_fixes={"WCS4432":{"userAgent":"(Mozilla/5[.]0 \\\\((iPhone|iPad|iPod);.*OS...
  14. Max

    "Is it possible to extract audio from a WebRTC voice call and apply real-time effects to the audio stream?"

    The server audio processing may be applied to streaming only, not to SIP calls. WCS is just a SIP gateway, not a PBX, so it may redirect media streams from browser to SIP PBX and vice versa. We recommend to process SIP audio at SIP PBX side (Asterisk, OpenSIPS etc).
  15. Max

    "Is it possible to extract audio from a WebRTC voice call and apply real-time effects to the audio stream?"

    Good day. You can add a custom Java class to intercept a decoded audio frames at server side: Server audio processing. Note that this feature requires audio transcoding to be enabled on the server, that increases the server CPU load in its turn.
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