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

[Rich Adamson]

> 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.

Cool!

How did it get set to 60ms?