I’m using JsSIP and react-native-webrtc for audio only communication. I’ve managed to start session and get remote and local stream objects. Our SIP server converts voip calls to a normal phone call. I know I successfully start session because I can call my own phone number and answer it.
Since I can’t use RTCView I can’t play audio. And since there is no audio transmitting, server gives a time out after a while and stops session, and phone hangs up. How can I start sending and receiving sound?
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I'm having a problem with react native
my app can let you video call each other using webRTC and socket.io, the technologies i use are WebRTC, React Native, Socket.io , socket.io-client, react-native-webrtc
Basically the way my app works is when you open the app you automatically connect to socket.io to listen and make calls but when I turn off the phone screen or exit the app I can't hear and receive calls can call again because at that time the socket has automatically disconnected.
I want my application can still work properly even if I exit the application my application can still receive notifications of incoming calls from another person, can say I want my application Works like Messenger App
My current workaround is to find a way for each user's socket to always be connected to listen for incoming calls from others.
Has anyone encountered this problem or have a solution for this please let me know, I really appreciate it
Thank you guys, Have a nice day <3
Move your socket to the Background service and then you can add the socket events in the service.
Note: This will increase your battery consumption.
I'm trying to isolate video and audio and am able to control the video feed from the caller side, however, unable to turn off the local video stream on the remote side since its an audio call. Any suggestions on how to isolate the video and audio feeds. It doesn't work just by removing the streams by getting the getStream.
I have extended tutorial one to one call for recording.
Original http://doc-kurento.readthedocs.io/en/stable/tutorials.html#webrtc-one-to-one-video-call
Extended https://github.com/gaikwad411/kurento-tutorial-node
Everything is fine but recording the remote audio.
When caller and callee videos are recorded, in the caller video recording callee voice is absent and vica versa.
I have searched kurento docs and mailing lists but did not find solution.
The workarounds I have in mind
1. Use ffmpeg to combine two videos
2. Use composite recording, I will also need to combine remote audio stream.
My questions are
1) Why it is happening, because I can hear the remote audio in ongoing call, but not in recording. In recording I can hear my own voice only.
2) Is there another solution apart from composite recording.
This is perfectly normal behaviour. When you connect a WebRtcEndpoint to a RecorderEndpoint, you only get the media that the endpoint is pushing into the pipeline. As the endpoint is one peer of a WebRTC connection between the browser and the media server, the media that the endpoint pushes into the pipeline is whatever it receives from the browser that has negotiated that WebRTC connection.
The only options that you have, as you have states already, are post-processing or composite mixing.
I have setup WebRTC using verto module of freeswitch and we have done everything successfully just the one thing we are experiencing since long time and could not get any solution.
I want to play a account balance when ever any user try to make call and if there is a no balance in account need to play "Not enough credit in account".
Now the problem is Freeswitch actually playing those notification which we can able to listen properly if we make call using soft-phone but i could not able to listen it via WebRTC web phone.
any one can help me to fix this issue?
I need to start a live stream in a remote computer connected to a webcam,
then connect to that remote ip address and see the live stream, like a security webcam more or less.
On my client i want to be able to see the stream in my browser.
What I've tried so far:
VLC on the remote pc: I start the stream (MMS, HTTP or RSTP) and then I encapsulate the stream as object in a html page.
This works, but I have a high latency and not all the browsers support x-vlc-plugin.
WebRTC. This seemed to me the best solution. Direct stream, very low latency.
I tried all the solutions I found in internet, that also integrate node.js. I tried also to build some code myself but the problem is that:
I start the stream on the "server", the remote pc.
When i go to the client, I type in the browser the ip address and port of the remote PC. In theory I should be able to see the REMOTE stream, but instead the browser asks for permission to use my LOCAL camera!
Do you have some hints or solutions about? What am I doing wrong?
Last project I tried:
https://github.com/xat/webcam-binaryjs-demo
In this project:
https://webrtc.github.io/samples/src/content/peerconnection/multiple-relay/
the developer uses a relay of the stream.
Buttons work but I don't know how to use this, that is how to catch the relay and display it on the client.
Thank you for your suggestions.
webRTC has three common API
getUserMedia : for communication and streaming between camera/mic with browser (request permission for access to camera/mic)
https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia
RTCDataChannel : data channel for send/receive any type of data on connection
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel
RTCPeerConnection : for creating peer-to-peer connection
https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection
you don't need getUserMedia
find getUserMedia() , this method send access request for camera and microphone to user , you can set both boolean false , or remove it carefully
navigator.getUserMedia({
video:false,
audio:true,
},function(mediaStream){...