webrtc

  1. I

    Прерывание трансляции на origin

    Здравствуйте. Версия - 5.2.891 ОС-AWS Linux При воспроизведении не наблюдалось скачка показателей производительности. Высылаю файл настроек, логи в форме отчета. Во время работы нашего приложения публикуемый поток начинает показывать черный прямоугольник. Происходит во время нагрузочного...
  2. M

    Video stream shows a white screen on Chrome 88 - Android 10

    Hello, Since yesterday we have started receiving some complaints of users reporting white screen in the video player on Chrome 88 - Android 10. Since it is a recent Chrome version, it could be a fresh bug but we could not reproduce it in a device using the same versions. We have many users on...
  3. I

    Высокие показатели nack count на origin

    Версия - 5.2.885 ОС - AWS Linux При публикации всегда растёт Nack count. В некоторых случаях 20/с, в некоторых 1/с. На edge при воспроизведении того же потока такой проблемы нет (проверял во вкладке media devices). Воспроизводится как для камеры, так и для рабочего стола. Chrome, Safari.
  4. M

    Live Stream getting freeze

    Hello Max We are having a issue of live stream freeze while streaming. Is there any config settings that we can make to overcome this live stream freeze issue? Something like increasing amount of: video_incoming_buffer_size Or any thing else than can help to fix this issue?
  5. Александр

    Логгирование событий веб приложения на сервер Flashphoner

    Здравствуйте, у меня возникла проблема при запуске веб-приложения, воспроизводящего трансляции видеофайла с диска, через сервер Flashphoner. На сервере ffmpeg преобразует видеофайл на диске в rtmp поток, с именем которое я указываю в опциях стрима в веб-приложении. Приложение использует WebRTC...
  6. M

    Delayed calls without human action

    Hi, we are receiving some complaints from our clients that assure that their apps are making calls without any action. We are using the webcallserver as gateway from the webrtc app in android to SIP. We initially thought that the problem was in the mobile app, but we finally reproduced it, and...
  7. E

    Прыгает картинка и звук при RTMP трансляции

    Доброго времени суток, проигрываю в плеере RTMP поток, который транслирую на сервер flashphonera через obs - https://docs.flashphoner.com/pages/viewpage.action?pageId=9241508#id-ВплеерепоRTMP-Известныепроблемы Проблема в следующем: картинка и звук прыгают/колеблются(приложил видеофайл для...
  8. S

    WebRTC player plays on mobile network but not on wifi

    Hi, I have a strange case that I can't reproduce at home on my setup but I have a user who when he is broadcasting while being connected to his home wifi I can't play his stream with WebRTC player, however, I can play the stream by fallback-ing to WSplayer from Flashphoner SDK. When this user...
  9. sabb

    RTMP Stream with error of Failed with ICE Timeout

    Error on playing a stream being published through ffmpeg for mp4 file The session shows as available though in player it fails with the above error. Server - ssl is configured with real domain name, site is configured with trial license from Flashphoner Able to test two way streaming through...
  10. alexey

    Низкая пропускная способность сети

    Добрый день! Поднял 2 сервера, для тестов на docker в Яндекс Облаке. Характеристики серверов примерно 32vcpu/96Gb памяти. Делал по инструкции https://flashphoner.com/kakoy-nuzhen-server-dlya-1000-webrtc-strimov/?lang=ru проверял сеть с iperf в результате 3,7Gbps. Однако во время теста видим вот...
  11. Dani

    can't open steam on iphone 11 and 12 while in a phone call.

    Finally figured out the issue with iphone11 and iphone12 - and it is reproduced with your own demo. You can't publish a stream while using the phone (regular call).
  12. appsgenii

    Flashphoner stop working

    My videos streams were working fine but it stoped today without any changes done on server. I have tried rtmp and rtsp both they are working directly on VLC but not from flashphoner. Getting this error when trying to stream from admin panal as well. FAILED Failed by ICE timeout Regards Adnan
  13. Dani

    Setting quality based on internet bandwidth

    Is there a way to change the stream quality if the internet bandwidth is good ? there is the target div dimensions, the camera resolution and the steam quality - is there a way to make a better "stream" when I know both sides has good internet connection ?
  14. K

    Долгое получение видеопотока 4К от камеры Axis

    Здравствуйте, Подскажите, какие настройки поменять на сервере или камере, чтобы видеопоток открывался быстрее. Сейчас при тестировании в примере client2/examples/demo/streaming/player/player.html после нажания кнопки Play видеопоток отображается через 20-25 секунд. Камера, сервер и плейер в...
  15. kip9696

    WebRTC is not visible in provider

    Hi! I am currently making an WebRTC player, If I create a player on the local server, WebRTC looks good But WebRTC is not visible when I upload the source to the server Please refer to picture below [local] [server] The source and connection environment of both are the same Answer please...
  16. S

    Publish not working on Firefox after update to FlashphonerWebCallServer-5.2.859

    Publishing does not work anymore after updating the WCS server version to 5.2.859. Steps to reproduce: 1. Use 2 way streaming example on demo 2. Click Connect button (result ok) 3. Click Publish button (fails) An error message appears: FAILED Browser error detected: The object can not be found...
  17. kip9696

    About the ports that need to be opened behind the firewall

    I am configuring a server behind a firewall udp range changed to 20000-25000 and service ports(ex 8080, 8888 more....) opened Mainly used module is rtsp to Webrtc But I cant receive webrtc video When all ports are open, I can receive Video But When closed except that port, I just receive...
  18. E

    Мониторинг статуса видео потока от участника конференции

    Доброго времени суток! Использую RoomApi для создания конференций, вот в чем вопрос. Бывают ситуации, когда видео от пользователя зависает намертво(остаётся только один кадр),при этом у того, кто этот поток транслирует, проблем нет. Если после зависания в конференцию зайдет еще человек, то будет...
  19. A

    SIP connectivity with opensips

    Hi there We are trying to make a connectivity with SIP API, we have successfully installed opensips on WCS and its running, kindly let us know how to define the sip extension on this and how to connect the web client? Thanks AB
  20. T

    Stream cut-off while recording on MacBook Safari

    Hi We've discovered an issue that seems exclusive to MacBook (Big Sur) Safari where the stream recording stops capturing video and audio after 8-10 seconds. Testing on the WCS Stream Recording demo page, we would do a recording of about 15 seconds. When playing back, the audio and video will...
Top