[Freeswitch-dev] Eavesdrop doesn't work for WebRTC calls

[Yan Brenman]

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 at jerris.com> wrote:

> how about on master?
>
> On Mar 15, 2014, at 10:19 PM, Yan Brenman <ybrenman at 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.
>
> Thanks
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