[Freeswitch-dev] Eavesdrop application doesn't work for the WebRTC clients

[Yan Brenman]

My apologies. I can this note on the Wiki now: "
Note: All testing of WebRTC should be done on the master branch from 
git. WebRTC is still very new and 1.4.beta branch probably does not have 
all the latest patches when you are testing"

Thanks again

On 3/21/2014 9:37 AM, Ken Rice wrote:
> Re: [Freeswitch-dev] Eavesdrop application doesn't work for the WebRTC 
> clients Where does it say this on the wiki? Webrtc is very new code 
> and while it is in the 1.4.beta branch, its best to test it on the 
> master branch where all the development happens... We should change 
> that fact on the wiki... If you can point me to the correct page I'll 
> see to it.
>
> K
>
>
> On 3/21/14 9:03 AM, "Yan Brenman" <yb91423 at gmail.com> wrote:
>
>     Hi Ken,
>
>     thank you very much for the reply. Actually, I am using branch
>     1.4.beta (which according to the Wiki is the branch
>     which provides WebRTC support) to test "eavesdrop" with the WebRTC
>     clients. I only merged some code from
>     video-media-bug branch to get the video working with "eavesdrop".
>     Without those changes only audio was going
>     working. But again, nothing is working for the WebRTC clients.
>     I will follow your suggestion and try the master branch master branch.
>
>     Thanks
>     Yan
>
>
>     On Fri, Mar 21, 2014 at 8:38 AM, Ken Rice <krice at freeswitch.org>
>     wrote:
>
>         You should be testing webrtc stuff on the master branch...
>         Theres been a lot of changes going on in tree.. The
>         video-media-bug branch is probably incomplete... I don't think
>         its even close to ready for merging at this point...
>
>
>         On 3/18/14 2:01 PM, "Yan Brenman" <yb91423 at gmail.com
>         <http://yb91423@gmail.com> > wrote:
>
>             Hello,
>
>             I was trying to get the eavesdrop application work on the
>             1.4.beta branch for the WebRTC clients (works great for
>             all other
>             types of clients) and could not. Neither audio nor video
>             (I merged the code from the video-media-bug branch for the
>             video)
>             is going through. For the client I tried both - sipml5 and
>             jssip clients.
>
>             Please let me know if there is any other piece of
>             information I can provide.
>
>             I will be more then happy to do any kind of code changes
>             and testing - just need some starting points. Especially
>             considering
>             that I am very new to the WebRTC.
>
>             Every piece of advice is greatly appreciated.
>
>             Thanks
>             Yan
>
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