Search results

  1. Max

    WCS service issue with AWS Markekplace AMI

    We cannot reproduce the issue starting a new instance from latest Marketplace AMI (5.2.944-systemd246) with your userdata script: WCS is started successfully without any permission fixes. Please provide SSH acceess to the problem instance using this form, we will check
  2. Max

    Возможность отслеживания ивентов mute (video/audio)

    Завели тикет WCS-3245 по бэкпорту данной функции в Android SDK 1.0. Сообщим здесь о готовности. Обращаем внимание, что на устройствах с Android 8 и выше необходимо использовать Android SDK 1.1.
  3. Max

    WebRTC Video not getting pushed

    No. There were some fixes affecting CDN subscribers, but you have problem on publisher side. There were also some H264 parsing fixes which affects streams publishing and recording, may be this helps.
  4. Max

    Fatal Exception: java.lang.NullPointerException

    Good day. We fixed a possible race condition in Android SDK build 1.0.1.78. Please update and check.
  5. Max

    no sound in recordings

    The problem is definitely in RTSP stream. But we cannot detect from logs on your server, does the synchronization value grows monotonically while playing the strem during a 36 hours, or does it jumps up occasionaly. So please provide access to the stream from outside networks for us to play it...
  6. Max

    WCS service issue with AWS Markekplace AMI

    Good day. Please try the following sudo /usr/local/FlashphonerWebCallServer/webcallserver set-permissions sudo systemctl restart webcallserver
  7. Max

    WebRTC Video not getting pushed

    Good day. Perhaps this is publisher channel issue. Please try to use more stable network for publishing (Wi-Fi instead of 4G, wired connection instead of Wi-Fi etc). WebRTC over TCP publishing may also help: session.createStream({ name: streamName, display: localVideo, ...
  8. Max

    Получить данные о просмотре трансляции

    Добрый день. Статус PLAYING выставляется потоку, когда установлено WebRTC соединение и поднята медиасессия для этого подписчика. При этом видео начнет играть на клиенте не раньше, чем будет получен первый ключевой кадр, содержащий основную информацию о видео на данный момент времени...
  9. Max

    Intermittently failed to play audio over webrtc/turn

    Please provide SSH access to the server using this form.
  10. Max

    Launching Flashphoner server in AWS Auto Scaling Group

    Yes. Yes, as described in the article. In the doc (p 2.3) just HTTP ports are shown. If you plan to use HTTPS, you should provide SSL certificates for load balancer domain name.
  11. Max

    Не корректное получение стирма на Android - Pixel 3 и Pixel 3XL

    Добрый день. В сборке 1.1.0.30 параметры картинки при публикации по умолчанию установлены в 320x240, 30 fps.
  12. Max

    Launching Flashphoner server in AWS Auto Scaling Group

    Yes, you can apply your current license. But in this case you should use custom image, not Marketplace AMI.
  13. Max

    Launching Flashphoner server in AWS Auto Scaling Group

    Good day. Please read this article with step by step guide. Also read the details in this doc. You can apply the same license key to all the instances. If you're using AWS Marketplace AMI, it will be billed by Amazon itself for their own price. If you're using custom base image with your own...
  14. Max

    Intermittently failed to play audio over webrtc/turn

    Please try to switsh off RTP bundle support rtp_bundle=false and add local interface to ICE candidates rtc_ice_add_local_interface=true Also please make sure you are connecting to WCS to play a stream with Force relay enabled: Flashphoner.createSession({ urlServer: url, mediaOptions: {...
  15. Max

    How to publish the Live RTSP URL and play in Chrome browser

    You can try to launch ffmpeg from your Java/.Net application: https://docs.flashphoner.com/display/WCS52EN/RTP+stream+publishing+via+RTSP ffmpeg -stream_loop -1 -re -i bunny360p.mp4 -c:a libopus -ac 2 -ar 48000 -c:v copy -b:a 96K -b:v 500K -f rtsp -rtsp_transport tcp...
  16. Max

    How to publish the Live RTSP URL and play in Chrome browser

    1. You do a SOAP request. 2. As a result you have RTSP stream URL (does this correct?). 3. You copy RTSP URL into the player and do play. Example of player: https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html
  17. Max

    Intermittently failed to play audio over webrtc/turn

    Good day. Please collect a full report as described here and send using this form. We will check.
  18. Max

    Возможность отслеживания ивентов mute (video/audio)

    Добрый день. В сборке Android SDK 1.1.0.29 добавлена возможность определить при проигрывании, заглушено ли аудио или видео в потоке. Подробнее здесь.
  19. Max

    no sound in recordings

    No, this is default value and does not affect the case We added the settings audio_incoming_buffer_size=100 video_incoming_buffer_size=100 Then we monitored synchronization values for 3 available streams and recorded the stream with hearable audio periodically using REST API. During the tests...
  20. Max

    Тормоза видео на HTTPS странице.

    TCP дает задержку. Если WebRTC по UDP дает задержку меньше 1 секунды при трансляции, то TCP может давать до 3 секунд. Проверьте два участка канала - между камерой и сервером и между играющим клиентом и сервером: 1. С одной стороны поднимаем iperf сервер (порт должен быть открыт) iperf3 -s -p...
Top