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 Fri, Mar 21, 2014 at 9:37 AM, Ken Rice <krice@freeswitch.org> wrote:
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@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@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@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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