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@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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