Re: [Linphone-developers] Bursty rtp traffic

[Simon Morlat]

Le Wednesday 20 May 2009 16:40:07 Vadim Lebedev, vous avez écrit :
> Simon,
>
> I'm not sure...
> We're using portaudio and we're tryed bot pa_mme backend and pa_dsound
> backed with similar results.
> What happens is that et the beginning of the stream we see packets
> spaced pretty regularily at 20 ms but later
> the situation degrade.
>
> I've impression that this is somehow related to regularity of ms_ticker..
The MSTicker does a very good job, however what could produce is a MSFilter (perhaps the portaudio ones?) that do blocking calls within their .process method, thus slowing the whole graph processing.

>
> We're digging into it.
>
> Thanks
> Vadim
>
> Simon Morlat wrote:
> > 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
> > > >
> > > >
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