webrtc

  1. A

    SIP connectivity with opensips

    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
  2. T

    Stream cut-off while recording on MacBook Safari

    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...
  3. fabiojapa

    Webm problem on Google Chrome

    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...
  4. djuka

    Automatic resolution switch

    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?
  5. djuka

    Limited FPS from mobile

    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?
  6. K

    About using webrtc from non-SSL web screen

    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...
  7. S

    call browser from mobile

    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
  8. A

    Playback of audio only stream is not working on latest build of Chrome 87

    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...
  9. richard-vd

    distorted audio on Apple devices

    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...
  10. K

    Firefox on iPhone

    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:
  11. vanarie

    Twilio group sceen to RTMP

    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...
  12. J

    Republishing edited video stream

    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?
  13. Al_Ku

    iPad не подключается к WCS

    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...
  14. L

    Отсутствие звука в iOS 14

    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.
  15. D

    Using Flashphoner to receive calls with "standard WebRTC"

    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...
  16. A

    Video stuck from Mobile connections

    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...
  17. M

    Publisher playback video black

    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...
  18. F

    Audio stream over RTSP problem on new system

    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...
  19. inSnat

    Failed to parse SessionDescription

    When using only audio it sends me the following error message, even in its own demo page in the Media Devices section it sends the same when not sending video Uncaught (in promise) DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to parse SessionDescription...
  20. A

    Webphonr to Sip phone connectivity

    Hi there We would like to connect peer to peer connectivity with the webphone to a sip phone and vice versa the clint sip phone is running on a LAN where WCS is running... The local sip phone is xlite and does not require any registration or user name... Kindly suggest what information we need...
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