[Asterisk-Users] Asterisk SIP Packet Time (20ms)

[Andres]

On Monday 22 December 2003 16:37, Andres wrote:
> On Monday 22 December 2003 15:36, Rich Adamson wrote:
> > > I have a question regarding the Asterisk Packet Time for SIP Calls.  It
> > > is hardcoded at 20ms but when I do an RTP Analysis on a stream it is
> > > clear that these packets are not spaced out at 20ms.  In general you
> > > see something like:
> > >
> > > Packet 50 - Delay 50ms
> > > Packet 51 - Delay 5ms
> > > Packet 52 - Delay 5ms
> > > Packet 53 - Delay 50ms
> > > Packet 54 - Delay 5ms
> > > Packet 55 - Delay 5ms
> > >
> > > Is there anyway to space them out evenly at 20ms??
> >
> > The 20 ms is not the inter-packet timing, its the relative content of
> > what's within the packet. In other words, the packet contains 20ms of
> > encoded voice.
> >
> > If the inter-packet times (delays) are large, as they would seem to be
> > in your example, then something else is not right. Possibly a half-duplex
> > ethernet connection, something else running on the server, router
> > buffers, etc.
> >
> > On a typical * --> C7960 local call, I generally see from 1ms to 20ms
> > inter-packet delays. Seldom (if ever) anything above 20ms.
>
> Thanks for your Input Rich.  I went ahead and tested this on our production
> servers and sure enough the inter-packet times are 20ms.  There must be
> something happening with our LAB Asterisk.  It could be the CBQ traffic
> shaping software we have running on it.  I will fiddle around with it to
> see if it changes anything.
>
> Thanks!
> Andres
Ok...after some more testing, the traffic shaping software was not the 
culprit.  It turns out that if the UA is configured for 60ms of voice, then 
Asterisk will show this strange behaviour.  If we set the UA for 20ms, then 
all works well.

Thanks.
Andres

>
> > Rich
> >
> >
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