[asterisk-dev] SDP interop on SFU

[Joshua Colp]

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