webrtc

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...

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 -...

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...

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...

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?

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

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...

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...

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

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...

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...

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...

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.

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...

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.