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

[Andres]

On Monday 22 December 2003 19:58, Rich Adamson wrote:
> > 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?
The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the 
transmit packet size to 60ms (or multiple other values).  Asterisk will 
receive 60ms and transmit 20ms times 3 packets, andit works quite well.  In 
any case our SPA2000 problem was unrelated to the packet time.

Regards,
Andres 
>
>
>
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