It can use either. By default, preference is given to UDP, but depending on the firewall(s) in between the peers connecting it may only be able to connect with TCP. You can use Wireshark to capture packets and verify whether TCP or UDP is being used. In Chrome you can also see details on the … Read more
You should definitely have a look at Kurento. It provides a WebRTC server infrastructure that allows you to record from a WebRTC feed and much more. You can also find some examples for the application you are planning here. It is really easy to add recording capabilities to that demo, and store the media file … Read more
jeffreyveon’s answer will help reduce the chance that your user will choose deny, since she will only have to choose once. In case she does click deny, you can provide a message that explains why you need the permission and how to update her choice. For example: navigator.getUserMedia ( // constraints { video: true, audio: … Read more
As it was pretty much covered here, what you are trying to do here is not possible with plain, old-fashionned WebRTC (strictly peer-to-peer). Because as it was said earlier, WebRTC connections renegotiate encryption keys to encrypt data, for each session. So your broadcaster (B) will indeed need to upload its stream as many times as … Read more
We have quite similar problem. So far the only workaround we were able to do is to remove the meta tag for it to be “apple-mobile-web-app-capable” and let users to open it in Safari, where everything seems to work normally.
EDIT Since this answer has been originally posted the browser API has changed. .stop() is no longer available on the stream that gets passed to the callback. The developer will have to access the tracks that make up the stream (audio or video) and stop each of them individually. More info here: https://developers.google.com/web/updates/2015/07/mediastream-deprecations?hl=en#stop-ended-and-active Example (from … Read more
WebRTC is designed for high-performance, high quality communication of video, audio and arbitrary data. In other words, for apps exactly like what you describe. WebRTC apps need a service via which they can exchange network and media metadata, a process known as signaling. However, once signaling has taken place, video/audio/data is streamed directly between clients, … Read more