We are facing a strange issue in webrtc call i.e. in a connected webrtc audio only call when some one upgrade the call(add video) the audio tracks will drops from orignator side.
steps to reproduce the problem
1.make a audio only call between two peers A and B.
2.updgrade call to video by calling getUserMedia again from peer A.
3.call established.
4.A can hear audio and view video.
5.B cant hear audio.
What is the expected result?
onaddstream(e) e.stream should contain both audio and video tracks
What do you see instead?
only video track is there at B's side (recipient)
What version of the product are you using? On what operating system?
Chrome 51/WIndows7
Please find the webrtc dump from below link
Webrtc dump
Related
I've been trying to debug this issue, I'm creating a peer to peer audio streaming app.
Each time a mobile device using Safari receives a peers incoming MediaStream, it plays the stream for a fraction of a second and then stops.
I've checked the actual MediaStream & Audio object I'm using to play the incoming stream, but nothing of its status suggests that's it's paused, inactive or stopped.
I know there are some issues with Safari when it comes to programmatically playing Audio, but I thought that was on the initial playback of the clients first stream and then all subsequent streams should work.
I used the implementation from this article.
Anyone know of the reason for this?
I am trying to record google meet audio and video using webRtc. I found mediaRecorder Api for same. But i am unable to record meeting, as it is only capturing my webcam and audio. How can i record a whole meeting through code in any language?
One solution is to use Neko to record conferences. It is like jibri but works for all any website!
It is a headless Chromium instance that runs on a server. You launch the neko instance and then join the conference call. You then can capture the web page's audio and video.
You can specify a RTMP output (Twitch, Youtube...) or you can save the files.
I'm new to Stack Overflow and I have a question: I want to make an online TV website. I've got the UDP streaming links for each TV channels (ex. udp://#225.1.2.249:30120). Then, I can't find the way how to play that stream on video player in browsers. (I've already tried Video.js and hls.js but both not work, but it easily play on VLC).
I know that my grammar is terrible, hope you forgive that.
You don’t. Browsers support HTTP, Websockets, and webrtc. Any other protocol requires a browser plugin.
I know how to access audio input devices via getUserMedia() and route them to the WebAudio API. This works fine for microphones and such. But in my use-case, I would rather like to hook into the audio stream of an output device. The use case is that I want to create a spectrum analyser for audio coming from a digital audio workstation (DAW) running on the same PC.
I tried to enumerate the devices and call getUserMedia() with the device id of an audio device, but the stream returned only showed silence data. The only solution I found so far is to install an audio loopback device (like Soundflower on Macs) to route the DAW's output to and then use this as an input device for getUserMedia(). But this will require the user to install 3rd party software.
Is there any way to hook directly into the audio stream of an output device instead, before it is actually sent to the physical device (speaker or external soundcard)?
This can be achieved using the desktop capture APIs (chrome.desktopCapture.chooseDesktopMedia). An example for chrome is included here
So I want to be able to send a normal video from a video file (AVI or any other) through WebRTC, can that be done? The only examples I see of WebRTC are video chats, so I feel as if its only geared towards webcam and chats.
So my question is, technically can sending normal video from a video file (not webcam) over WebRTC be done?
Try: "Pre-recorded media streaming" --- Documentation and Source Code.
This experiment uses MediaSource API to render Blobs in <video> element. This experiment has some issues need to be fixed e.g. it can't send longer WebM videos.
You can try this experiment as well.
The codecs typically used in AVI are not directly supported by WebRTC clients, but if you are writing your own standalone client then of course it could read an AVI or other video file and transcode it to VP8 video and Opus audio (or whatever other codecs you were able to negotiate), and transmit it via RTP. If you are trying to do video transcoding in JavaScript in a browser then that will be very slow.