Hey Alexandre,

> What version of SFLphone are you running? Is this problem occurs between two 
> SFLphone instances? Is the stream audio stopped in both way 
> (incoming/outgoing), or in only one way?
>   

Version 0.9.9. It might be only incoming but of course now it's not
doing it.. :-) I'll check next time it happens.
My SFLphone talks to my Asterisk box that talks to other Asterisk boxes
that talk to PSTN phones via VOIP providers or to hardware VOIP phones.

> When putting the phone on hold and taking it back, you reinitialize the RTP 
> session and thus reinitialize the RTP local timestamp. I know that our RTP 
> stack will, by default, consider late RTP packets as expired and thus drop 
> them. So my first guess would be that the RTP stack is not processing fast 
> enough the incoming data packets.
>
> On way to solve this would be to implement the 
> OST::RtpSession::onPacketReceive method callback in 
> src/audio/audiortp/AudioRtpSession.h . This may require some refactoring of 
> the AudioRtpFactory as it use templates, which change the callback function 
> signature.
>   

I'm unable to help modify the code at the moment but will gladly test
patches.

I haven't yet had a change to do the debugging (not so many calls on
Sunday :-) ).... does the debug output include any stats on expired RTP
packets?

thanks and regards,
Jan

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