search for: getusermedia

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

...et to the server* > > My questions are: > 1. Is wss now required by sipml5 live demo (implying wiki page is not > up-to-date) ? > > Yes, like the error says, you have to use wss on pages served via https. > Furthermore, Chrome requires the use of https when you want to use > getUserMedia. > See here: > https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. > It says: " Starting with Chrome 47, getUserMedia() requests are only > allowed from secure origins: HTTPS or localhost." > Is it implied here that both HTTPS and WSS must also come f...

hello, now that html5 audio becomes usable in some browsers, for example http://webaudiodemos.appspot.com/AudioRecorder/index.html ... has anybody thought about to implement a web based source client for icecast? that's for now beyond my coding skills, unfortunately... thx, uno

On 05/03/2014 04:40 PM, Brad Isbell wrote: > I have built a web-based source client using the Web Audio API and > getUserMedia, called the AudioPump Web Encoder. It uses the SHOUTcast > source protocol for now, but I have the code nearly ready for the > Icecast source protocol for when someone needs it. > > You can see a demo here: http://demo.audiopump.co:3000/ That's pretty impressive. > The Web E...

Following up on this topic ( sorry if this starts a new thread but I just joined the ml ), I do no understand why it is not possible to use the audio stream from webRTC's getUserMedia and then send it over a websocket ? It seems that the webRTC implementation can natively encode in ogg format in stereo from any interface ( according to https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC ). Why wouldnt it be suitable ? In the other, I assume, like Jamie, that t...

...Z6OT_LbWKMhRVJ34nO--ntbwhfeOr2eQPuLJFoCCVr1EJPw at mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > Hi Brad, > > 2014-05-03 11:40 GMT-05:00 Brad Isbell <brad at musatcha.com>: > > I have built a web-based source client using the Web Audio API and > > getUserMedia, called the AudioPump Web Encoder. It uses the SHOUTcast > > source protocol for now, but I have the code nearly ready for the Icecast > > source protocol for when someone needs it. > > > > You can see a demo here: http://demo.audiopump.co:3000/ > > > > The Web...

I have built a web-based source client using the Web Audio API and getUserMedia, called the AudioPump Web Encoder. It uses the SHOUTcast source protocol for now, but I have the code nearly ready for the Icecast source protocol for when someone needs it. You can see a demo here: http://demo.audiopump.co:3000/ The Web Encoder is built with a jQuery Mobile UI optimized for mo...