Matthieu LAURENT wrote:
> 
> Philippe Kalaf wrote:
> 
> Hi,
> 
> Could you tell me what you have been thinking about as a begin of solution?
> Is the solution based on RTCP (for example Sending Report)?
> Do you know when you will have a beta version? I can help you to test it 
> maybe.
> Will your solution work for VoIP application only or for streaming too?
We have recently made some design work to solve many issues this one
included. I will do some documentation about it soon and you will be
able to see how it should look like. As for the actual code, we will
work on it whenever we can but I can't say when any of it will be finished.

> 
> For your information, I have also asked the same question to the gstreamer 
> mailing list, and I have been answered that I should place my 2 pipes in one 
> pipe.
> I don't think this will enable to have a correct control of the 2 buffer 
> jitters.

The idea for lipsync is to have bins that have all streams from a common
SSRC and does the sync for us.

Best Regards,
Philippe
> 
> Regards
> 
> Matthieu
> 
> 
> 
> 
> 
> 
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