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  1. Max

    record problem question

    Yes. In this case, there should be a message in server logs Problem to move file /tmp/stream-ca00d7d0-52c2-11ee-ba8e-e9beb144939a-94brlh973bo9nlpe228922hort.mp4 to /usr/local/FlashphonerWebCallServer/records/stream-ca00d7d0-52c2-11ee-ba8e-e9beb144939a-94brlh973bo9nlpe228922hort.mp4 and...
  2. Max

    record problem question

    Good day. WCS does not check any network conditions when writing stream recordings to disk. It even does not know what the disk is: hard drive or network mounted drive. Linux file system should care about it. So temporary files are always created in record_tmp_dir if the folder exists and...
  3. Max

    Video stopped by the browser

    Please provide a links to test. Please look at Player example using custom volume slider and Embed Player example using standard controls.
  4. Max

    Video stopped by the browser

    There is no a solution without a real user action. You can provide some landing page where user has to interact with your site or click something before playback.
  5. Max

    Video stopped by the browser

    This is autoplay policy issue. The user action is required to unmute video: click a button or move a volume slider. You can use the method Stream.unmuteRemoteAudio() when click event occurs. Also, in last Web SDK builds you can enable standard video controls using the stream option useControls...
  6. Max

    Video stopped by the browser

    Good day. This is Chrome autoplay policy issue. Video element should be muted to play video automatically. Please update Web SDK to the latest build 2.0.233 (or via NPM) and set the following stream option var options = { ..., unmutePlayOnStart: false }; var stream =...
  7. Max

    Need to change rtmp alias "live" to another

    Please open the file ~/.ssh/config: nano ~/.ssh/config and add the following to the file Host 127.0.0.1 HostkeyAlgorithms +ssh-rsa PubkeyAcceptedAlgorithms +ssh-rsa This should fix the issue.
  8. Max

    Need to change rtmp alias "live" to another

    You should add an application per every alias: live, live2 etc.
  9. Max

    Need to change rtmp alias "live" to another

    Please replace alias in the command example to your own alias (flash if you need)
  10. Max

    Кодек WebRTC

    Пожалуйста, воспроизведите проблему и соберите отчет на стороне сервера, как описано здесь, включая дебаговые логи и дамп трафика: Формирование отладочных логов и дампа трафика. Используйте скрипт для упаковки отчета: Сбор отладочных логов при помощи скрипта report.sh. Отправьте архив при помощи...
  11. Max

    Need to change rtmp alias "live" to another

    Good day. Please clarify: do you want to publish an RTMP stream to WCS using URL like rtmp://wcs:1935/alias/streamName? If yes, you should do the following: 1. Enter to WCS CLI: Using and accessing command line 2. Add a new server application to use for RTMP streams publishing: add app -m...
  12. Max

    Encrypted SSL Certificate file

    Hello When you upload SSL files, WCS does import these files into single file /usr/local/FlashphonerWebCallServer/conf/wss.jks Even if you upload Encrypted SSL files, you would need to upload decryption password too. Therefore Upload works with un-ecnrypted SSL only. As you can see, there is no...
  13. Max

    Unable to see rtsp stream, but stream correctly open in vlc

    We can debug this, but we need an access to the camera to debug. If you cannot provide the access, please use the workaround.
  14. Max

    Unable to see rtsp stream, but stream correctly open in vlc

    This is not strange, it's a normal behaviour: if the stream is protected by username and password, both of them should be provided.
  15. Max

    Unable to see rtsp stream, but stream correctly open in vlc

    Good day. Please provide us a public RTSP link to test using this private form, we will check. Seems like you've not set a valid username and/or password to play RTSP stream.
  16. Max

    MSE broken since FlashphonerWebCallServer-5.2.1408

    We found the issue reason and testing the fix. Also, there is a simple workaround you can use with latest builds: set the following parameter rtp_session_init_always=true and RTSP streams should play via MSE. When we release the fix, you can revert the setting back.
  17. Max

    MSE broken since FlashphonerWebCallServer-5.2.1408

    Good day. We confirm the issue when playing pulled RTSP streams (but all the other streams work) via MSE. Raised the ticket WCS-3909. Will let you know about fix. The only workaround is to roll back to 5.2.1404.
  18. Max

    SIP video call from WCS can't hear audio of audio file was play from SIP server

    Good day. We're already testing the fix. Please be patient.
  19. Max

    I want to do voice recognition.

    Good day. At server side, you can implement Java class to get audio tracks in PCM format and redirect it to third-party voice recognition tools: Server audio processing At client side, you can access audio track via video tag: stream = session.createStream(options).on(STREAM_STATUS.PENDING...
  20. Max

    Disable 8081

    Good day. If you want to disable HTTP port but use only HTTPS 8444, you can set up firewall (iptables, fwd etc) or NAT router to block requests to port 8081 from outside, for example iptables -A INPUT -p tcp --dport 8081 -j DROP If you want to disable some actions on HTTP interface, please refer...
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