Re: [Linphone-developers] Bursty rtp traffic

[Simon Morlat]

It can also be due to waveApi buffers that are larger than 20 ms, as a consequence several packets are delivered through the capture callback at the same time.
Linphone does this also on windows. This must not be a problem thanks to jitter buffer.

Simon

Le Tuesday 19 May 2009 17:01:54 Petr Kuba, vous avez écrit :
> Hello,
>
> I guess that this is caused by Windows timer which is not enough precise
> to send packets each 20ms. E.g. if you try to sleep() for times smaller
> than 10ms the result is very poor - sometimes it waits for too long time
> and sometimes it does not wait at all. Therefore it is typically
> necessary to use longer delays.
>
> Regards,
> Petr
>
> Vadim Lebedev wrote:
> > Hello,
> >
> > We're trying to use mediastreamer2 in QuteCom and it mostly works well,
> > but sometimes
> > we're seeing strange behaviour:
> >
> > When using G711 codec, adaptive jitter control, 20 msec packet size...
> > I see bursty RTP output: Each 60 ms there is burst of 3 rtp packets.
> > We're using portaudio and it's on windows/xp
> >
> > Any ideas?
> >
> > Thanks
> > Vadim
> >
> >
> > _______________________________________________
> > Linphone-developers mailing list
> > address@hidden
> > http://lists.nongnu.org/mailman/listinfo/linphone-developers