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

Yan Brenman ybrenman at xopnetworks.com
Tue Mar 25 06:15:18 MSK 2014


And just for completeness - eavesdrop on the bridged audio call doesn't 
work either.

On 3/24/2014 10:41 AM, Yan Brenman wrote:
> 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 at gmail.com 
>> <mailto:yb91423 at 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 at jerris.com 
>>> <mailto:mike at 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 at xopnetworks.com
>>>     <javascript:_e(%7B%7D,'cvml','ybrenman at 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 at 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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>>
>>
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>
>
>
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> http://www.freeswitchsolutions.com
>
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