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

[Andres]

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

>
> Rich
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users