[Asterisk-Users] Asterisk SIP Packet Time (20ms)
andres at telesip.net
Mon Dec 22 14:37:20 MST 2003
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,
> 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.
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