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> 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

------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

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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@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

------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <http://consulting@freeswitch.org> http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com

Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com

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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

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I just tried the "master" branch and situation there with the "eavesdrop" application is absolutely the same as it is on the "1.4.beta" branch. It doesn't work for the WebRTC clients and it does work for the non-WebRTC clients (I used jitsi client to test). Having said that I was only getting audio working with "eavesdrop" for non-WebRTC clients but I am pretty sure if I merge in some code I used to from the video-media-bug branch - I will get the video too. Just to describe you my test scenario - I am placing and inbound leg with I am "answering" and "park". Then from the FS_CLI I am "originating" an outbound leg which is "eavesdrop" on the inbound "parked" leg. I hope you don't see anything wrong with it. Thanks again On 3/21/2014 8:38 AM, Ken Rice wrote:

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

------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com

Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com

FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

-- Ken _http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _irc.freenode.net #freeswitch Twitter: @FreeSWITCH

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com

Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com

FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

Sorry, it's answered and parked. And I expect to hear audio and see video, as i do with non-WebRTC client (jitsi for example). On Monday, March 24, 2014, Michael Jerris <mike@jerris.com> wrote:

What do you expect to be hearing when eavesdropping on a parked channel?

On Mar 21, 2014, at 9:05 PM, Yan Brenman <ybrenman@xopnetworks.com<javascript:_e(%7B%7D,'cvml','ybrenman@xopnetworks.com');>> wrote:

I just tried the "master" branch and situation there with the "eavesdrop" application is absolutely the same as it is on the "1.4.beta" branch. It doesn't work for the WebRTC clients and it does work for the non-WebRTC clients (I used jitsi client to test). Having said that I was only getting audio working with "eavesdrop" for non-WebRTC clients but I am pretty sure if I merge in some code I used to from the video-media-bug branch - I will get the video too.

Just to describe you my test scenario - I am placing and inbound leg with I am "answering" and "park". Then from the FS_CLI I am "originating" an outbound leg which is "eavesdrop" on the inbound "parked" leg. I hope you don't see anything wrong with it.

Thanks again

On 3/21/2014 8:38 AM, Ken Rice 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> 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

I understand the situation with the video (although like I said - I did merged code from video-media-bug branch and got video working for non-webrtc clients). Eavesdropping on the answered and parked call leg from WebRTC client with audio only gives the same result - no audio on the eavesdropping leg. Although, according to the log, everything went ok without any problems. I compared the SDP session dump from the creating eavesdropping leg to the dump where I would just "uuid_bridge" two webrtc call legs - and there are no differences there (like I thought the eavesdropping leg would somehow end up using the wrong cocec or something). I have not tried to eavesdrop an already bridged call as ideally this is not the scenario I would be planning to use - I don't want caller leg (the one we are eavesdropping on) to get terminated if call leg it is bridged to hangs up. And I have not tried the recording yet. Will do it next. Thanks On 3/24/2014 7:10 AM, Michael Jerris wrote:

I'll leave the video question totally alone as thats code that isn't even merged in yet, and still very under development. What happens when you eavesdrop a bridged call with audio only? Does call recording behave differently?

On Mar 24, 2014, at 9:58 AM, Yan Brenman <yb91423@gmail.com <mailto:yb91423@gmail.com>> wrote:

...

Sorry, it's answered and parked. And I expect to hear audio and see video, as i do with non-WebRTC client (jitsi for example).

On Monday, March 24, 2014, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote:

What do you expect to be hearing when eavesdropping on a parked channel?

On Mar 21, 2014, at 9:05 PM, Yan Brenman <ybrenman@xopnetworks.com <javascript:_e(%7B%7D,'cvml','ybrenman@xopnetworks.com');>> wrote:

...

I just tried the "master" branch and situation there with the "eavesdrop" application is absolutely the same as it is on the "1.4.beta" branch. It doesn't work for the WebRTC clients and it does work for the non-WebRTC clients (I used jitsi client to test). Having said that I was only getting audio working with "eavesdrop" for non-WebRTC clients but I am pretty sure if I merge in some code I used to from the video-media-bug branch - I will get the video too.

Just to describe you my test scenario - I am placing and inbound leg with I am "answering" and "park". Then from the FS_CLI I am "originating" an outbound leg which is "eavesdrop" on the inbound "parked" leg. I hope you don't see anything wrong with it.

Thanks again

On 3/21/2014 8:38 AM, Ken Rice 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> 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

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com

Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com

FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com

Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com

FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org