webrtc

https://demo.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html не работает в Opera (Version:70.0.3728.178 на Linux) Есть ощущение, что Опера в отличие от других браузеров пытается использовать flash, увы даже его включение не помогает начать воспроизведение.

Hi, we're using an almost year old version of the sever/ api. the reason is that whenever we tried to upgrade - iPhone video is frozen. No issues on MAC and no issue on android. I can also confirm that the OTHER side see the "frozen" video as live - the problem looks local to iphone (trying...

Hi, I've installed WCS 5 in a AWS instance. Purchased the $75 monthly subscription. Added the licence to the instance. When I log on to the https://ip_address:8888 everything is fine - the two way streaming works fine. I now have created a broadcaster in another AWS instance. But when I try...

We have recently setup ICE server which is different from our existing WCS server. We decided to go for TURN server as web rtc was failing in some networks intermittently. When I try to create a media session with that ICE server, it times out saying failed by ICE timeout. But on the TURN server...

Hi, I am using an FFmpeg command to push the mp4 to the flashphoner. I am using below FFmpeg command I can play the stream via VLC player using the rtmp endpoint, but when I try to play the stream using the Webrtc from Flashphoner demo player it is not working. I see this screen For...

we get many complaints recently about session disconnections. Is there a way to lower the threshold - so session will not disconnect with every small lag ? Thanks.

Hi Community! We are now working 2 month on a website in order to be able to stream multiple channel through a web Call Server 5. Our solution needs to be compatible with most of the last gen browser like announced by FlashPhoner. We were about to buy a full licence but we are still...

Hello and thanks in advance for reading, newbie here. Of all the player options I feel more comfortable with the iframe embed because I couldn't get the other ones to work (I tried, and I recognize I must be doing something wrong). Well, currently I'm using this: <iframe...

При попытке пользователем получить стрим с сервера возвращается ошибка Other: No free ports available Также при просмотре потоков через веб-морду

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...

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...

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

Использую функцию захвата экрана с последующим выводом изображения в плеер. Захват работает стабильно, а на плеер изображение выводится от случая к случаю. При этом настройки никакие не меняю. Просто то работает, то нет.

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?

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...

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.

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.

My source is an RTSP stream with stereo AAC audio. I want to achieve high quality stereo audio encoding in the direction of server to browser. The AAC and Opus bitrates are configurable in flashphoner.properties, but it's still mono. How can I set the Opus and AAC encoders to stereo? And it...

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.

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...