1. andrew.n

    Video distortion when changing RTCEAGLVideoView size

    I use 2 RTCEAGLVideoViews, one to play the remote link and one for broadcasting. When the streamers are ready, the delegate method - (void)videoView: (RTCEAGLVideoView *)videoView didChangeVideoSize: (CGSize)size; is called and I set the videoView.layer.frame properly. We have the following...
  2. M

    WebRTC Streaming issues

    Hi, I have two issues with webRTC Streaming, 1) After sometime when a stream is published, the video goes black at streaming side but at viewer side, the stream can still be viewed. Please check attached image below 2) Streaming gets disconnected automatically after 1-2 hours. I found below...
  3. M

    Error with WebRTC stream

    Hi, I am recieving following error in origin server logs ERROR BitstreamNormalizer - STUN-UDP-pool-46-thread-253 Failed to normalize SPS...
  4. KonstantinK

    буффер на стороне клиента

    при воспроизведении потока в VLC видео идет с задержкой, но не замирает а при воспроизведении потока в HTML5 плеере видео идет без задежки, но замерает. подскажите есть ли возможность использования буфера на стороне клиента для HTML5 плеера?
  5. domi91c

    WebRTC to RTMP: Reducing latency

    I recently setup WCS5 on an AWS micro instance. My web app allows two users to have a conversation through WebRTC, and I'm sending one side of that conversation to Flashphoner to be converted in real-time to an RTMP stream, which I then feed into OBS as a Media Source. From the web app to OBS...
  6. R

    AbstractStunSocket - FScheduling-pool-44-thread-2 Can not find local candidate

    Hello! Once we attempt to add a participant to mixer / mcu, we got: 20:43:54,083 WARN AbstractStunSocket - FScheduling-pool-44-thread-2 Can not find local candidate for /X.X.X.X We are using Google Cloud.
  7. R

    Call Duration

    Hello! There is an way to receive back from hooks or from API the duration of a CALL / ROOM? For example, Room A is started at: 12:00:00 and Room A is closed at: 12:35:00 -> Duration of this room: 35 minuts.
  8. 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.
  9. 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...
  10. 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 -...
  11. 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" and there are no ICE candidates present. This is similar to the playStream example documented at...
  12. 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...
  13. 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?
  14. 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 ?
  15. 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...
  16. 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...
  17. A

    Не работает стрим

    Переодически при попытке запуска стрима, сервер выдает ошибку, после перезагрузки проблема решается, файл с логами прилагаю ниже
  18. 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...
  19. 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...
  20. 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...