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

[Rich Adamson]

> 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.

Rich