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  1. Max

    Ошибка сертивикатов

    Мы такой импорт не тестировали. Тестирование может занять время. Наиболее быстрый способ - установить HA Proxy, который проксирует websocket трафик на WCS по адресу ws://localhost:8080 И на HA Proxy уже сконфигурировать поддержку двух доменов. Т.е. WCS работает по ws (http), а HA Proxy...
  2. Max

    Stream Restriction by authorize domains or ip addresses?

    Currently WCS does not support domain restriction. We will inform you once it is implemented. For authentication, you can generate temporary short-living tokens and pass the tokens to play() method. You can also authorize by IP address. Example: 1. You play a stream...
  3. Max

    Ошибка сертивикатов

    Можно импортировать из командной строки: https://flashphoner.com/docs/wcs5/wcs_docs/html/ru/wcs-admin-guide/index.html?security-ssl_certificates_management-websocket_ssl-ssl_certificate_import.htm Или дождаться фикса с обновлением, если это баг. В какой версии сервера проблема?
  4. Max

    Not enough bandwidth

    Yes, then NOT_ENOUGH_BANDWIDTH event will be raised when loss reaches 10%.
  5. Max

    Not enough bandwidth

    webrtc_cc2_local_k_threshold setting is obsolete, it has been replaced with webrtc_cc2_bitrate_overuse_event_threshold. By default (i.e. if the setting is not specified in config flashphoner.properties), NOT_ENOUGH_BANDWIDTH event is raised when loss reaches 5%...
  6. Max

    Обрыв стримера при rtmp

    попробуйте выставить в настройках OBS тип кодирования ultrafast была подобная проблема без этой настройки
  7. Max

    websocket is working, webrtc not

    Could you provide SSH access to the server and access to the WCS dashboard? If yes, please send to logs@flashphoner.com.
  8. Max

    Обрыв стримера при rtmp

    Нужно отключить keep alives для RTMP Еще тема по настройкам keep alives
  9. Max

    websocket is working, webrtc not

    Can the issue be reproduced when Wowza is not running? I.e., stop WCS, stop Wowza and other services which could use the port range, and start WCS.
  10. Max

    Not enough bandwidth

    Could you provide access via SSH to the server and access to the WCS dashboard? If yes, please send to logs@flashphoner.com.
  11. Max

    SIP as RTMP issues

    There is a configuration issue: codecs not to be used in SIP as RTMP case should be listed in 'codecs_exclude_sip_rtmp' not removed from 'codecs'. For example, codecs =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv codecs_exclude_sip =mpeg4-generic,flv,mpv...
  12. Max

    adding custom watermark on video stream?

    Yes it is possible. You can intercept a transcoded stream by stream name and add watermark to your transcoded stream. This can be done on server-side. 1. Create Java class. package com.example.video; import com.flashphoner.sdk.media.*; import org.slf4j.Logger; import org.slf4j.LoggerFactory...
  13. Max

    adding custom watermark on video stream?

    Hello You can try to set custom_watermark_filename=mylogo.png in flashphoner.properties and place mylogo.png file in WCS_HOME/conf directory. The mylogo.png file should be 640x480 The watermark will not be displayed without transcoding. Therefore 1. You have to force video transcoding, setting...
  14. Max

    Какие порты нужно оставить открытыми, чтобы принудительно пустить webrtc via tcp?

    То есть сервер и клиент на одном и том же компьютере? Вы не могли бы дать доступ к серверу и dashboard? (Выслать можно на logs@flashphoner.com.)
  15. Max

    websocket is working, webrtc not

    The stream has resolution 1280x720, video doesn't appear because there is not enough bandwidth. It can be played on WCS as WebRTC if specify video constraints for the stream to decrease playback resolution (checked with the latest v. 2429). var constraints = { audio: true, video: {...
  16. Max

    websocket is working, webrtc not

    Generally we do not check this email. Please use logs@flashphoner.com for any emails related the forum. We found sent RTSP URL. It does not play in VLC for me. Please check on your end. This is result of ffprobe ffprobe rtsp://host:1935/stream ffprobe version 3.2.3 Copyright (c) 2007-2017 the...
  17. Max

    SIP as RTMP issues

    Please provide following logs and files: /usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log /usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner_manager.log /usr/local/FlashphonerWebCallServer/conf and pcap file of traffic dump tcpdump -s 4096 -w log.pcap You can...
  18. Max

    websocket is working, webrtc not

    Please provide RTSP URL of the stream.
  19. Max

    Какие порты нужно оставить открытыми, чтобы принудительно пустить webrtc via tcp?

    Проверили пример Firewall Streaming (firewall-traversal-streaming) на Ubuntu 16.04 (c TURN-сервером turn.flashphoner.com и последней версией WCS - 2428) - работает и с Firefox 55, и с Chrome 60 (с "iceTransportPolicy": "relay" и без). Как проверялось с Firefox?
  20. Max

    Публикация с строоних приложений.

    Попробуйте передать appKey: rtmp://localhost:1935/live?user=123&password=123&appKey=streamFlash appKey - это зарезервированный параметр, который должен распознаваться.
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