webrtc

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...

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...

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

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...

I am having some problems on stream recording of webm video. The record is ok, but after some time the problem start: The webm is recorded, but don't play in Google Chrome. The same webm file works fine in Firefox. I need to restart de Flashphoner WCS, and the problem is solved. But after some...

Hi, I have set video resolution like this: video: { width: {min:320,max:640}, height: {min:240,max:480} } On mobile devices I get recordings always 640x480 even if the network is slow. Is there a possibility to automatically switch from 640x480 to 320x240 if the network is slow?

I try to use FP to record streams from mobile but from different mobile devices, I got recordings with different framerate regardless of what is set in the javascript constraints. I set 25 for frameRate but recordings have form 10 to 30. What is the purpose of frameRate property actually?

Hello, Please tell me about using webrtc from non-SSL web screen. I'm trying to create a player for webrtc in a web app. Since this web application is used within the LAN, it will be executed under HTTP. (Example: http: // <ServerIP> /webrtc.html web server uses apache) As a test, I ported...

Hi, I saw this article: https://flashphoner.com/call-from-browser-to-mobile/ I was wondering whether the opposite is possible as well. Can I call a browser from a mobile phone? best regards Piotr

Hi, Our users are reporting after updating the chrome, our audio-only stream has stopped working. Getting connection error message "Failed by DTLS error" Any ideas? I have updated chrome on dev machine and its same error. Its quite urgent if you can help. Our server is running on 5.2.768...

In recent releases of Apple software (macOS Big Sur using Safari 14.0.1, iOS 14.2 and iPadOS 14.2) the audio of my RTSP stream starts heavily distorted when using WebRTC (Opus audio). After some time audio becomes clean but it can go back to the distorted state later on. I already tried changing...

Hello, Does Web SDK work in Firefox on iOS? In "Streamer" example we get "FAILED: Local error" message (see attachment), and in our project we get the following error:

Hello, I've been testing with WCS and setup a test server. I was able to get webRTC -> WCS -> RTMP working from the install and read the rtmp://myserver stream with VLC media player. I assume that's a good test. My goal now is to find a good, scalable way to port a video meeting conference...

I'm building an app to use webrtc to capture video from web browser, stream it as RTMP stream, edited the RTMP stream using ffmpeg then forward it to FB or Instagram. To add to above, I want to re-publish the edited video stream as a new RTMP stream...how do I do that in Flashphoner/webcallserver?

Здравствуйте! Имеется свежая установка WCS v.0.5.28.2753-5.2.820-df1a5f7314be94770e0c91156ce32181913a22bb. Установлен на VPS CentOS 7. Демонстрационная лицензия. Демонстрационные приложения в Chrome в CentOS и Android открываются и работают нормально (тестирую two way streaming). При открытии...

Возникла такая проблема, при публикации WebRTC потока с телефона, а именно iOS 14 + Safari отключаются все звуки на сайте для паблишера, а также отключаются звуки нажатий клавиш (тапов по клавиатуре), системный звуки уведомления етц. Как только публикация прекращается всё возвращается в норму.

Hi, I'm not very knowledgeable about WebRTC or Flashphoner, so please bear with me. My company got an open source call software in C# that enables us to make WebRTC calls in Unity. To do that, I need to fill the following information: (and with this information, the call works) However, when...

Hi, We have observed that in video chat if the caller is joining from mobile phone connection the picture gets stuck probably due to the low bandwidth, kindly assist how we can avoid this scenario.. our javascript settings for both sides are as following: conn.join({name...

We started to receive complaints from a few customers on Android and iPhone with black video when publishing. Our connection workflow is the following: 1. Broadcaster open the page, video playback is shown and it connects to Flashphoner server to create the session (Video showing) 2...

hello i have issues with the definitive system where I lose audio once In flashphoner. It worked with the demo on another system but not anymore with the definitive one. I can read the stream with video and audio in VLC but I lose audio in flashphoner on your demo page or on my system . I can...