On Wed, Mar 7, 2018, at 5:24 AM, Luca Pradovera wrote: > Hello, > that is a good starting point, thanks. > We are using SIP.js, and actually our client is just a modified version of > CyberMegaPhone2k. > > What happens is that video stream sent from a Chrome user, when received on > Firefox, behave in an inconsistent way. Very rarely, they work. Most of the > times, we get either: > - No video at all > - A few frames at the start, then freeze > - Working video with very bad quality
Then I'd suggest also using about:webrtc to examine the receive stream, as well as low level Firefox debug[1]. As the browsers are a black box problems may not end up appearing on the Asterisk side, or even the Javascript console leaving you to guess what is going on (and sometimes guessing wrongly). [1] https://gist.github.com/ibc/3a10b27812d99c8abd1b -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
