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

[Rich Adamson]

> > > Packet 50 - Delay 50ms
> > > Packet 51 - Delay 5ms
> > > Packet 52 - Delay 5ms
> > > Packet 53 - Delay 50ms
> > > Packet 54 - Delay 5ms
> > > Packet 55 - Delay 5ms
> 
> > 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 that is the case, then what is in packet 52 and 55?  There's not enough 
> time between packets for 20ms of voice, unless it's repeating audio in the 
> packets...

In this short example, if you add up all of the times shown and divide
by the number of entries, you'll see its exactly 20ms of voice. The two
delays of 50ms each are the problems (obviously).  All that short trace
really says is during the 120ms sampling interval, there was no dropped
udp packets (since they nicely add up). If they didn't add up, it would
be a problem.

So, the question really is... what caused the two 50 ms delays?

In technical network terms those delays are called "jitter", and in the
case noted above, that jitter is substantial.  If that trace was taken
next to the equipment (no routers involved in the middle), then which
ever piece of equipment that originated those packets should be looked
at carefully.

If doesn't make any difference if anyone was talking for not; the packets
are still going to flow, and they should be flowing at a very constant
rate with reasonable but constant inter-packet delay (jitter).

There really is nothing more that can be said without additional detail.

Rich