Thank you very much for the reply. Yes, I just got another reply suggesting to try master branch. The reason I was working with 1.4.beta branch is because that's what Wiki is suggesting. I am going to do this now. Thanks Yan On Fri, Mar 21, 2014 at 8:00 AM, Michael Jerris <mike@jerris.com> wrote:
how about on master?
On Mar 15, 2014, at 10:19 PM, Yan Brenman <ybrenman@xopnetworks.com> wrote:
Hello,
I have been running the 1.4.beta branch of the FreeSWITCH to do some WebRTC related testing. And I noticed that "eavesdrop" application, although working nicely for the any other types of clients, doesn't work for the WebRTC. I am not getting neither video nor audio (I did merge changes from the video-media-bug branch and do get video "eavesdropped" nicely using jitsi clients) using SIPML5 clients.
I will be more than happy to do any code modifications and testing required, just need some initial input as I am relatively new to WebRTC. I looked at the current implementation of "eavesdrop" and everything seems to make perfect sense. Why it doesn't work for the WebRTC then?
Please let me know if there is anything I can provide.
Any help is greatly appreciated.
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