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Re: [Linphone-developers] Bursty rtp traffic

From: Vadim Lebedev
Subject: Re: [Linphone-developers] Bursty rtp traffic
Date: Wed, 20 May 2009 16:40:07 +0200
User-agent: Thunderbird (X11/20090318)


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..

We're digging into it.


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.


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
> >

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