webrtc

  1. Schimek

    RTMP to WebRTC (conference - room)

    Hi, Is it possible to inject (display) an rtmp stream in a conference room? I saw the example rtmp to webrtc, unfortunately there is no talk of conferences. Thank you.
  2. V

    Flasphoner webrtc not playing in the Firefox.

    When I try to play the webrtc stream in Firefox it is not playing. The status in the player remains Established. I added the config rtc_ice_add_local_interface=true as mentioned here https://forum.flashphoner.com/threads/webrtc-stream-not-publishing-in-firefox.12921/#post-23041. Still, it is...
  3. J

    Треск в звуке у участника видеоконференции

    Здравствуйте! У одного из наших участников появляется некий треск в момент проведения видеоконференции между несколькими пользователями. Могли бы уточнить с чем это может быть связано и на что обратить внимание? Пример на видео -...
  4. C

    ICE candidates not sent by client

    How are the ICE candidates sent from the browser client to the server when using WebRTC? On the WebSocket, the playStream message is sent with "c=IN IP4 0.0.0.0" and there are no ICE candidates present. This is similar to the playStream example documented at...
  5. sebastien

    Firefox issue

    Hi I build a conference call and its work in chrome, opera, safari, samsung internet but in Firefox i have some issues. First of all i have secured the server with an ssl key. The problem that I have is when I use Firefox and I join an existing room the session disconnect often. It reconnects...
  6. burak guder

    what is the codec of the Screen Sharing ?

    I want to share screen with re-publish (rtmp-push) feature but only h264 codec can be used because the v8 cod requires extra core usage. How do I ensure that only h264 is used in webrtc inputs?
  7. A

    Артефакты на видео

    https://api.eyezon.app:8444/client/records/5f072aa60bad5a09d2d4ec33-5edf51f21db4c23abcf85ec5_1f820fa3-2f4b-42a5-9a80-ae92a9dbde66.webm Почему на видео могут быть такие артефакты ?
  8. richard-vd

    freezes (but only when WCS output is UDP)

    My stream often freezes for a few seconds, while audio continues uninterrupted. It only happens when WCS outputs UDP, either directly using UDP WebRTC or even indirectly using the internal TURN server (in that case the only use of UDP is between TURN and WCS!). The same stream over TCP WebRTC or...
  9. J

    Stream is getting failed/disconnected

    Hi Max, suddenly our stream started getting disconnect after every minute or two, please suggest it's a prod issue 13:13:12,806 INFO WSServerHandler - WS-pool-19-thread-214 Orgign: null 13:13:12,807 WARN WSServerHandler - WS-pool-19-thread-214 Close channel [id: 0x74bb5297...
  10. A

    Не работает стрим

    Переодически при попытке запуска стрима, сервер выдает ошибку, после перезагрузки проблема решается, файл с логами прилагаю ниже
  11. paulishuku

    New to this, trying to stream my RTSP network feed.

    I've been following the tutorial on how to add my network camera to the stream via RTSP. I can connect to my external rtsp link via VLC, and so can others outside my network who have tested so i know the port forwarding is working. However, when i launch the aws instance and connect through the...
  12. A

    Перестали рабоать стримы

    Сегодня с утра перестал работать сервер, как ваш демо на вашем сайте, так и у нас, ругается на невозможность подключения Ошб:14 http://ppa.launchpad.net/heyarje/libav-11/ubuntu xenial/main amd64 Packages 404 Not Found [IP: 2001:67c:1560:8008::15 80] ... Ошб:22...
  13. andrew.n

    Call Kit and Flashphoner

    Context: We have to add a new feature to our app, to support video streaming calls between 2 users. Same as Skype/Messanger but only for 2 users (no group support yet) As I understood, both Skype and Messanger use Call Kit to properly handle communication between multiple apps that supports...
  14. R

    Multiple Players at same page

    Hello! Its possible to use the player (webrtc) not embed, on the same page, to monitor arround 50 streams? Actually we are using this with the embed code and works.
  15. pirks

    Optimal cloud instance

    I am building a streaming system based on your server. The wcs server receives 100 rtsp streams at the input. At the output, I want to distribute 50 streams using webrtc and HLS (80% / 20%) .The total number of viewers is not more than 300. Also 50 streams should be recorded as HLS. The question...
  16. T

    Help with None of MediaProviders available

    I set up a laravel website with a self hosted player but I'm getting this error: None of MediaProviders available But when put the codes to a raw .html page, it works.
  17. angel

    Change display html element on the fly

    Hi, We are creating a html page that display multiple RTSP to WEBRTC streams. We need to change the html display element option when the stream is running but we cannot figure it out how to do it. Is there any way to change the display html element option once the stream is running? Thanks a lot!
  18. SlavaCh

    streaming from ReactNative app

    Hi We experience black screen instead of stream while seting the react native app and tried with iframe in the app. Iframe is not loaded from the phone (iOS 13.5.1) when using WebView and loading html page and the link to the stream itself...
  19. R

    WebRTC - User to user stream

    Hello! Is it supported use WCS server and user to user stream, to reduce server load?
  20. R

    PULL - PUSH

    Hello! We are trying to PULL stream from another WCS server using API. We got 200 CODE return, but stream is not being played. Server A -> StandAlone -> User connect and stream webcam Server B -> PULL stream from server A. How to? We follow the documentation but not works.
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