Thanks for updating SVN. This means that we can now build straight from the
source stream, which is very nice. Your blind shot at a fix was quite
accurate! :-)

As for the jitter buffer, it would be nice to do some work to completely
remove any chance of reproducing the 'silence issue' but unfortunately I
don't think I can do that at the moment. I have found that with these
changes I can no longer delay the RTP stream long enough to cause a problem
so it should get by for a while.

>From my perspective these recent changes make sipXtapi a really useful tool
for building SIP UA applications. It will be interesting to see what
difference the media reworking makes.

Thanks,
Alex

-----Original Message-----
From: Jaroslav Libak [mailto:[EMAIL PROTECTED] 
Sent: 30 July 2007 17:42
To: Alexander Boreham
Cc: sipxtapi-devlist.sip...
Subject: Re: [sipxtapi-dev] Windows queue replacement for mic/speaker

Alexander Boreham wrote:
> Thank you very much for this patch. I applied it and have tried minimizing
> windows and other actions and it works perfectly.
> 
> I had to manually add OsIntPtrMsg.cpp into the sipXportLib project but I
> think you expected this to be the case.
> 
> We could reproduce this minimizing windows problem in the ezPhone. The
> problem manifested as skipping whilst minimizing or maximizing windows.
> However after doing it a dozen times or so the sound would cut off
> completely and would only come back upon another window operation. From
that
> point onwards the sound would toggle on and off with each windows
> minimize/maximize operation.
> 
> This patch corrects the problem with minimizing of windows on Windows :-)
> 
> Many thanks yet again,
> Alex

I applied the patch to SVN. For me it was a blind shot, as I didn't
encounter any audio problems when minimizing windows.

If you plan to use sipxtapi in a client, then you also need to replace
the jitter buffer. Maybe I will replace it with speex jitter buffer if I
have the time.

Jaro
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